- sr-users - kamailio.org

[Users] dbtable
by jammers yann 16 Jan '06

16 Jan '06

Can i have the role of each table in openser database ? ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

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[Users] OpenSER TLS with Windows Messenger
by Cagri Koksal 16 Jan '06

16 Jan '06

Hi all, I'm trying to test openset TLS feature with windows messenger. I setup openser on Debian Linux and modified the openser.cfg and openssl.cnfas required by the TLS faq and docs at openser.org. When I try to sign in using windows messenger I got the following message from OpenSer: "tls-accept: Error in SSL:" At the same time windows messenger gives the following error messenger; "There was a problem verifying the certificate from the server ....." Here is an excerpt from my openser.cfg - disable_tls = 0 listen = tls:192.168.0.128:5061 tls_verify = 0 tls_require_certificate = 0 tls_method = TLSv1 ( I also tried SSLv23 ) tls_certificate = "/usr/local/etc/openser/tls/sipsrv_rootCA/sipsrv01/cert.pem" tls_private_key = "/usr/local/etc/openser/tls/sipsrv_rootCA/sipsrv01/privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/sipsrv_rootCA/sipsrv01/calist.pem" and an excerpt from my openssl.cnf dir = "/usr/local/etc/openser/tls/sipsrv_rootCA/demoCA" certs = $dir/certs crl_dir = $dir/crl database = $dir/index.txt new_certs_dir = $dir/newcerts certificate = $dir/cacert.pem serial = $dir/serial crl = $dir/crl.pem I also used the existing certificates that come with openser-tls package. It resulted the same. Any suggestion or guidance is appreciated, thanks. -Cagri Koksal

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[Users] Radius Accounting
by martin＠campus-merseburg.de 16 Jan '06

16 Jan '06

Hi, why does OpenSER sends three Start-Radius-Packets on an Invite und and two on a bye? My config has these entries: [...] modparam("acc", "radius_flag", 3) [...] if (method=="BYE" || method=="CANCEL") { setflag(3); }; [...] } else if (method=="INVITE") { setflag(3); [...]

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[Users] TM functions : merging
by Bogdan-Andrei Iancu 15 Jan '06

15 Jan '06

Hi everybody, I just committed on CVS head some changes into TM script functions - related functions are merged into new ones; the goal is to simplify the usage of these functions and to open way for NAPTR lookup. Following functions were merged: - t_relay_to_udp(), t_relay_to_tcp(), t_relay_to_tls() merged into t_relay(proto:host:port); - t_forward_nonack_uri(), t_forward_nonack_udp(), t_forward_nonack_tcp(), t_forward_nonack_tls() merged into t_forward_nonack(proto:host:port) and t_forward_nonack(); - t_replicate_udp(), t_replicate_tcp(), t_replicate_tls() merged into t_replicate(proto:host:port) Note that the functionality is not affected. To make the transition easier, the replaced functions are not removed yet, but marked as obsolete (via documentation and messages at runtime). The plan is to have them removed soon - a second set of changes into the API is to come. regards, bogdan

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[Users] route on reply reason in failure_route
by Walter Schober 15 Jan '06

15 Jan '06

Can anyone please give me a hint how to access the reply reason in failure_route - or is it really not possible to trigger some event on the response _reason_. 0(704) SIP Reply (status): 0(704) version: <SIP/2.0> 0(704) status: <404> 0(704) reason: <CLIR Inactive> <----- I want this! There is a pseudo variable for that, but no function can search the replied reason! Is it really true, that you can use the reply status only (t_check_status("404"))? Using avpops the pseude variables $rr and $rs are <null> at that moment. Do I miss something? I'm doing that: failure_route[3] { if (t_check_status("404") ) { avp_printf("$reason", "The reply: $rr, $rs"); log("\n----------------------------------------\n"); avp_print(); log("\n----------------------------------------\n"); } } resulting in: 0(740) DEBUG:avpops:print_avp: p=0xb6177838, flags=2 0(740) DEBUG: id=<32> 0(740) DEBUG: val_str=<The reply: <null>, <null>> Any hints welcome. /Walter

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[Users] Problem with IAX and Openser.
by david.castro 13 Jan '06

13 Jan '06

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header ERROR: new_t: no valid From in INVITE ERROR: t_newtran: new_t failed ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Do you know the reason? Is this a bug? Which is guilty, asterisk or openser? I have this problem only in this scenario. Thank you. -- -------------------------------- David Castro Mosquera Adianta Tel: (34) 902 999 178 Fax: (34) 902 501 437 Email: david.castro(a)adianta.net Web: www.adianta.net --------------------------------

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[Users] New functionality : Direction detection in ACC module
by Bogdan-Andrei Iancu 13 Jan '06

13 Jan '06

Hi everybody, I just committed on CVS head a new functionality that was requested by Lenir <lenirsantiago(a)yahoo.com> some time ago. It's about detecting and compensating in the ACC records of the direction flow for sequential requests. For more details about it and about how to used it, please see the online documentation: http://openser.org/docs/modules/1.1.x/acc.html#AEN625 Any testing/feedback is highly appreciated ;). regards, bogdan

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[Users] delete specific contacts from location
by Mike Williams 13 Jan '06

13 Jan '06

Is it possible to delete a specific contact location for a user, and by using openserctl? I see that adding and deleting a certain contact is on the openser roadmap, and if this means that this functionality has made it into openser but not into openserctl, than I'm willing to try to add it myself. Any advice where to start is appreciated. Mike Williams

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[Users] Functionality
by Andréa Giordanna 13 Jan '06

13 Jan '06

Hi all I need a functionality on openser. The functionality would allow users to save on the server - a message, - a time and date - and a group of uris, and at the given time, the server would send the message to each one of the group of uris. This functionality would be useful for users to send reminders of events to a list of uris. Are there any modules on openser that do something like that? If not, how could I implement it? Thanks Andrea Giordanna --------------------------------- Yahoo! doce lar. Faça do Yahoo! sua homepage.

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[Users] about sems
by unplug 12 Jan '06

12 Jan '06

Since I have no idea about asterisk, and I would like to ask can I use sems instead of asterisk to provide same function. Any openser+sems howto available for fedora core4?

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[Users] Openser as a Outbound Proxy
by david.castro 12 Jan '06

12 Jan '06

This is the sip.conf from asterisk: ########################################################################### [general] bindport=5062 bindaddr=0.0.0.0 ;canreinvite=no disallow=all ;allow=all register => asterisk:asterisk@my_openser_ip ;you can found more info of this in http://www.voip-info.org/wiki/ [ser] allow=ulaw allow=alaw allow=g723.1 allow=g726 allow=g729 allow=gsm allow=ilbc allow=lpc10 allow=speex allow=adpcm type=friend ; We allow incoming and outgoing calls. Use peer if you are only doing MWI context=internal ; This is the context incoming calls land in host=ip_ser ; This is the hostname or IP address of your SER server ;fromdomain=ser.server.rld ; This is your SER_DOMAIN insecure=very ; This allows incoming calls from the phones routing through ser to be passed into asterisk dtmfmode=rfc2833 This is the openser.cfg ###################################################################### # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls #tls_verify = 1 #tls_require_certificate = 0 #tls_require_certificate = 1 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_certificate = "/usr/local/etc/openser/tls/openser1/cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_private_key = "/usr/local/etc/openser/tls/openser1/privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" #tls_ca_list = "/usr/local/etc/openser/tls/openser1/calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" #loadmodule "/usr/local/lib/openser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ xlog("yo soy <$si> y el destino es <$tu> <$td> <$du> <$rm> \n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # ----------------------------------------------------------------- # Handle NOTIFY requests from Sipura's to keep-alive NAT # ----------------------------------------------------------------- if ((is_method("NOTIFY")) && (! uri=~"sip:.*[@]+.*")) { xlog("entre en notify \n"); sl_send_reply("200", "OK keep-alive"); return; } # ----------------------------------------------------------------- # Only allow known sip messages,ack, pq no hace falta responder. # ----------------------------------------------------------------- if (!is_method("INVITE|REGISTER|BYE|ACK|CANCEL")) { xlog("Estoy en metodo no diponible y que es <$rm>\n"); sl_send_reply("501", "Not implemented here"); return; }; # 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()) { xlog("entree en loose route\n"); # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; # 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:*.") { if (method=="ACK" && (src_ip!=XXXXXX)){ # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXXXX", "subscriber")) { www_challenge("XXXXXXXX", "0"); exit; }; consume_credentials(); xlog("Voy a mandar respuesta a un ACK, yo soy <$si> y el destino es <$tu> <$td> <$du>\n"); route(1); exit; }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXX", "subscriber")) { www_challenge("XXXXX", "1"); exit; }; xlog("Me llego un register y yo soy <$si> y el destino es <$tu> <$td> <$du>\n"); consume_credentials(); save("location"); exit; }; if (method=="INVITE") { # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXX", "subscriber") && (src_ip!=XXXXXX) ) { www_challenge("XXXXX", "0"); exit; }; # consume_credentials(); xlog("Un tio mando un invite <$si> y el destino es <$tu> <$td> <$du> \n"); sl_send_reply("100", "Trying"); lookup("aliases"); #Si es de asterisk y no encuentra el telefono digo que no esta. if (!lookup("location") && (src_ip==XXXXXXXX) ) { #xlog("no encuentra el destino\n"); sl_send_reply("404", "Not Found"); exit; } else if (lookup("location") && (src_ip==XXXXX)){#if a call come from * and sip phone online i route to ring sl_send_reply("180", "Ringing"); route(2); return; } else {#otherway i forward to *. append_hf("P-hint: usrloc applied\r\n"); #sl_send_reply("180", "Ringing"); route(1); return; }; }; if (method=="CANCEL" && (src_ip==XXXXX)) {#Si viene de asterisk, lo mando con estado al telefo. #xlog("Me llego un cancel de asterisk\n"); route(2); return; } else if (method=="CANCEL" && (src_ip!=XXXXXX)) { #Si no viene de asterisk... # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXXX", "subscriber")) { www_challenge("XXXXXX", "0"); exit; }; consume_credentials(); #xlog("Me llego un cancel de NO asterisk\n"); lookup("aliases"); if (!lookup("location")) {#primero compruebo que este numero existe. #xlog("no encuentra el destino\n"); sl_send_reply("404", "Not Found"); exit; } else { route(1); #Se lo mando a asterisk. return; }; }; if (method=="BYE") { # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXX", "subscriber") && (src_ip!=XXXXXX) ) { www_challenge("XXXXX", "0"); exit; }; consume_credentials(); xlog("Me llego un BYE \n"); if (src_ip!=XXXXXXX){ route(1); } else { route(2); }; sl_send_reply("200", "OK"); exit; }; }; } #I route to asterisk here route[1] { xlog("Estoy en route(1) \n"); forward(XXXXX,5062); return; } #I route from asterisk to sip phones here route[2] { xlog("Estoy en route(2)\n"); if (!t_relay()) { #xlog("Dentro de error del t_relay del 2\n"); sl_reply_error(); }; return; } ################################################### Notes: XXXX=ip_asterisk This configuration works in the same net. Now, I'm trying to work behind a nat. This configuration don't works behind a nat. I don't challenge asterisk, if a call comes from asterisk_ip, I trust. Bye.

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[Users] NOTIFY and RFC
by Helge Waastad 12 Jan '06

12 Jan '06

Hi, My question today is more regarding RFC's. I've tried to find out if NOTIFY and Event: message-summary needs a Contact header. At least my Cisco gateway discards Notify's without a Contact header (400 Bad Request - 'Malformed/Missing Contact field') Is this a Cisco bug or SHOULD/MUST a NOTIFY have Contact of the sender? br hw

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[Users] Re: Users Digest, Vol 8, Issue 23
by unplug 12 Jan '06

12 Jan '06

As I know, each table is assigned to a module. For example, avpops module will use table usr_preferences. The usage of this module can be found in the module document. > Date: Wed, 11 Jan 2006 16:20:58 +0100 (CET) > From: jammers yann <foudelinux(a)yahoo.fr> > Subject: [Users] dbtable > To: users(a)openser.org > Message-ID: <20060111152058.85120.qmail(a)web25806.mail.ukl.yahoo.com> > Content-Type: text/plain; charset=iso-8859-1 > > Can i have the role of each table in openser database ? >

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[Users] web interface
by jammers yann 11 Jan '06

11 Jan '06

hi! do we have a web interface for openser like serweb for ser ? Or can we easily integarte serweb with openser thanks ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

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[Users] Re: OpenSER and transport selection (SRV and NAPTR)
by Greg Fausak 11 Jan '06

11 Jan '06

Howdy guys, I completely missed this thread. How is this going? I just read the roadmap which indicates NAPTR lookup is on the list. I've been maintaining NAPTR and SRV records looking forward to the day when t_relay() forwards using NAPTR (and SRV). Does that mean that the t_relay() will maintain the state, slogging through the NAPTR/SRV records appropriately, without me having to do additional logic??? Specifically, I'm keen on the SRV serial forking. -g Hi Bogdan, > I meant t_reply() will keep its behaviour as looking into URI for the > destination - but it will incorporate the NAPTR lookup. Great - this answers my question. > to response also to on earlier question, regarding the timing > - I say 3 months are more than enough ;). Excellent. Many thanks for your help (and patience ;) --Joachim > Joachim Fabini wrote: > > >Hi Bogdan, > > > > > > > >>indeed, there was a misunderstanding :) .t_relay() with no > >>param will be kept with the current functionality. > >>Or you suggest to be able to specify only a different proto > >>or port from the RURI? > >> > >> > > > >I did not yet find the primitive which is supposed to > >statefully relay and do the following: > >1. NAPTR query on the transport to use PLUS > >2. _exactly_ what t_relay() does now for the > > transport returned by the NAPTR query. > > > >You say that t_relay() is kept with the current functionality. > >Does this mean no NAPTR, or will t_relay() be extended to use > >NAPTR before SRV/A query? If the latter then everything is OK. > > > >Otherwise we have the alternatives: > > > >t_relay() -> Stateful relaying according to destination > > address using the incoming transport (no NAPTR). > >t_relay_to() -> Stateful relaying to a specific host (you said > > that <host> is mandatory here) using NAPTR to > > determine the transport. > > > > > >To summarize: > >My point is that none of these two primitives covers the case > >when the message is to be relayed to the next hop based only > >on the destination address _and_ using the transport selected > >by the destination proxy (determined via NAPTR query). > > > >Except if either a) t_relay() does NAPTR or b) the host > >parameter in t_relay_to() is optional. > > > >--Joachim > > > > > > > > > -- Greg Fausak greg(a)thursday.com

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[Users] Re: TLS problem
by david.castro 11 Jan '06

11 Jan '06

Thank you, I pleased with you if you could explain how I can implement that, or where I can found information about it. Thanks.

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[Users] Openser as a Outbound Proxy
by david.castro 11 Jan '06

11 Jan '06

I don't know if this help you, but I try. I have a similar scenario. I have openser, which deals with sip registers. If a sip phone make a call, I send the request to asterisk. Asterisk dispatch to PSTN or go back to openser (if this call is for sip). Asterisk deal the dialplan and cdr-csv and it's a gateway to PSTN. This is an idea. If you want i can post the config files. Bye. -- -------------------------------- David Castro Mosquera Adianta Tel: (34) 902 999 178 Fax: (34) 902 501 437 Email: david.castro(a)adianta.net Web: www.adianta.net --------------------------------

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Re: Re: [Users] Help: how to use pa module
by Kenny Yeh 11 Jan '06

11 Jan '06

samuel,hi yeah,as you said I changed cfg file,but ser is still cann't be started. Did you try? ======= 2006-01-11 16:43:02 您在来信中写道：======= >First of all: pa module is not stable and might not work. > >The other thing is that handle_subscription only accepts registrar or >jabber as parameters, so you have to change debian for registrar: > > handle_subscription("registrar"); > > >Luck!!! >Samuel. > >2006/1/11, Kenny Yeh <kenny(a)artdio.com.tw>: >> Hi,all >> >> I complied pa module in the right directory,but as the help file I cann't start openser right.my cfg file is below: >> >> #I added pa.so file >> loadmodule "/usr/local/lib/openser/modules/pa.so" >> # >> modparam("pa", "default_expires", 3600) >> handle_subscription("debian") >> >> debian is my main domain >> >> what's wrong with it? I need a help !!!! >> >> >> Kenny yeh >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-11

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[Users] Help: how to use pa module
by Kenny Yeh 11 Jan '06

11 Jan '06

Hi,all I complied pa module in the right directory,but as the help file I cann't start openser right.my cfg file is below: #I added pa.so file loadmodule "/usr/local/lib/openser/modules/pa.so" # modparam("pa", "default_expires", 3600) handle_subscription("debian") 　 debian is my main domain what's wrong with it? I need a help !!!! Kenny yeh

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[Users] Help: presence funtion
by Kenny Yeh 11 Jan '06

11 Jan '06

Hi,all I have a question: OpenSER can be a presence server? how ? I try to test by msn5.1,but OpenSER cann't recive 'subscribe' request. Can I load pa module to help that? BR! Kenny

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[Users] Via branch parameter
by Scott MacGregor 10 Jan '06

10 Jan '06

Is there a way to change the branch parameter in any of the Via headers in an INVITE? I am getting a 482 loop detected which I think is due to the same branch parameter in two requests. Thanks, Scott __________________________________________ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com

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[Users] tracking down problems adding permanent usrloc record
by Mike Williams 10 Jan '06

10 Jan '06

On Monday 02 January 2006 11:11, Juha Heinanen wrote: > Mike Williams writes: > > I created the account 11112223333(a)mydomain.net to test with, and then I > > attempted to use: > > > > openserctl ul add 11112223333(a)etcvoip.net sip:18122228203@etcvoip.net > > > > but it returned: > > > > error: 400; check if you use aliases in OpenSER > > > > I added the alias_db module to my config, but I still get errors. Any > > advice? Should I add them manually to the location database? > > i have never used openserctl. my management application writes > permanent contacts directly to location table. remember to tell via > flags field that registration is permanent. > > -- juha I modified the openserctl script to give more output: 400 table (aliases) not found error: 400; check if you use aliases in OpenSER on `openserctl ul add 11112223333(a)etcvoip.net sip:18122228203@etcvoip.net` I traced this message back to the ul_show_contact function in ul_fifo.c. Everything is fine until it gets to line 573, where it calls fifo_find_domain, which sets d=0. Then, in line 575, the error message given above is returned because d if now false. static inline void fifo_find_domain(str* _name, udomain_t** _d) { dlist_t* ptr; ptr = root; while(ptr) { if ((ptr->name.len == _name->len) && !memcmp(ptr->name.s, _name->s, _name->len)) { break; } ptr = ptr->next; } if (ptr) { *_d = ptr->d; } else { *_d = 0; } } I'm not a C programmer, so I don't really understand what's happening here, but it appears to be looping through a list to find a domain, checking to be sure their lengths and values match. It keeps looping until it finds one, and then quits on the first one. It appears that ptr doesn't point to anything in my case; it fails the next if, setting the passed-by-reference d to 0, leading me to think that it hasn't found any domains, or couldn't get a domain list in the beginning. What does this "ptr = root;" mean? I then tried running `openserctl domain show`, only to get: 500 command 'domain_dump' not available Now, I checked the actual database openser.domain table, and my domain appeared correctly. I am also loading the domain module on openser startup. Any ideas? I realize this is getting awfully deep into code, but I hope someone can help. It's probably a simple error on my part. Thanks in advance. Mike Williams

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[Users] LCR module
by Ovidiu Sas 10 Jan '06

10 Jan '06

Hi all, The new lcr module has now two modes of operation controlled by the direct_db parameter: - direct_db = 1 (no DB caching) This is the old behavior of the lcr module - direct_db = 0 (DB caching - default one) In this mode, the 1.7 TODO issue from the README file is addressed: no more SQL query in load_gws() function. In addition to this, the From URI from the lcr table is now a POSIX regular expression, with the following extension: if the From URI is an empty string, it will be considered as a universal match - ie .* without invoking the regexec function. Migrating from the old mode to the new mode requires the from_uri column from the lcr table to be updated to use POSIX regular expressions instead of the special _ and % characters. Example: old from_uri -> new from_uri ------------------------------ % -> 123% -> 123.* regards Ovidiu Sas

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[Users] TLS problem
by David 10 Jan '06

10 Jan '06

Hello. I'm trying to use TLS with openser. I've made the steps from the tutorial:http://openser.org/docs/tls.html. I have "tls_verify = 1" and "tls_require_certificate = 1". I have a phone without TLS support, but it registers in openser. I think, this shouldn't happend. Anybody has ideas? Second question: I used netstat to verify if somebody is listening on 5060 and 5061, but if I put ip_openser:5061 in the phone, I have an icmp message with "destination unreachable". What's happening? Thanks

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[Users] LCR problem
by Pepe 10 Jan '06

10 Jan '06

Hello, Im making tests and its not a LCR problem, its a problem from my GW2, when I use it for first option, it fails too, here you have the ngrep, ClientA --> Proxy --> GW2 (192.168.10.93) (192.168.10.91) (195.219.74.166) Regards The problem is that the BYE request will be handled by your LCR logic. The BYE request should be route in the loose_route block as it is an in-dialog request. Maybe the BYE sent from the gateway is not correct. Please post a ngrep dump (ngrep -t -W byline port 5060) regards klaus Pepe wrote: > Hello, > > Im configuring Openser with LCR module and Im having an extrange > behavior, I have 2 gateways, GW1(preference1) and GW2(preference2), > > GW1(pref.1) > / \ > ClientA --> OpenSer --> Client B > \ GW2 (pref.2) / > > > When I call from Client A to Client B using GW1, all works fine, its the > same when hang up Client B or Client A, but when GW1 fail(I provoke it > changing codec) and use failure route (GW 2) then if Client A hang up > all works fine, but the problem is when is Client B who hang up, its > like a new conversation, GW 2 send BYE to openser and Openser just send > "503 Service Not avilable - No gateways" to GW2, but doesnt send nothing > to ClientA, any idea ???? > > > Thx in advance >

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[Users] direct manipulation of DB
by Norman Brandinger 09 Jan '06

09 Jan '06

For example, using PHP, I have found that the code below works well: $a1="$subscriber:$domain:$password"; $ha1=md5($a1); $a1b="$subscriber@$domain:$domain:$password"; $ha1b=md5($a1b); Regards, Norm

2 1

[Users] problem accounting of ACKs
by Klaus Darilion 09 Jan '06

09 Jan '06

Hi! I have problems accounting ACK (e2e ACKs after INVITE-200 Ok). I have the following setup: client1 proxy1 proxy2 proxy3 client2 |---INVITE--->| | | | | rewritehostport | | | | t_relay | | | | |--INVITE->| | | | | enum_lookup | | | | t_relay | | | | |-INVITE-->| | | | | lookup() | | | | t_relay | | | | |-----INVITE->| | | | | | | | | |<----200 OK--| | | |<--200 OK-| | | |<--200 OK-| | | | <--200 OK---| | | | | | | | | | ---ACK----> | | | | | loose_route | | | | |----ACK-> | | | | | loose_route | | | | |---ACK--->| | | | | loose_route | | | | |---ACK------>| | | | | | All proxies do accounting. I do see the INVITE and BYE requests accountend by all proxies, but the ACK is only accounted by proxy1. All proxies are configured nearly identical: # accounting parameters modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 0) modparam("acc", "multi_leg_enabled", 0) modparam("acc", "failed_transaction_flag", 4) # radius, syslog and DB modparam("acc", "log_flag", 1) # syslog modparam("acc", "db_flag", 2) # DB modparam("acc", "db_missed_flag", 3) # DB missed calles table modparam("acc", "swap_direction", 1) and all the flags are set for sure: route{ setflag(1);setflag(2); setflag(3);setflag(4); Does somebody had a similar problem? Are there any known issues how ACKs are handled? Studying the debug log. it looks like proxy 1 identifies the ACK as belonging to a transaction, whereas proxy 2 does not find the transaction (full debug at the end of the mail): proxy 1: ----------- t_lookup_request: start searching: hash=6242, isACK=1 parse_headers: flags=38 DEBUG: t_lookup_request: e2e proxy ACK found parse_headers: flags=8 DEBUG: totag for e2e ACK found: 0 DBG: trans=0x406b1940, callback type 4, id 0 entered parse_headers: flags=258 ACC: request acknowledged: method=ACK, i-uri=sip:user201@..... proxy 2: ----------- t_lookup_request: start searching: hash=6242, isACK=1 DEBUG: proceeding to pre-RFC3261 transaction matching DEBUG: t_lookup_request: no transaction found DEBUG:tm:t_relay: forwarding ACK statelessly DEBUG: mk_proxy: doing DNS lookup... parse_headers: flags=1000 check_via_address(10.10.0.41, 10.10.0.41, 0) Sending: ACK sip:user201@10.10.1.50:8737 SIP/2.0^M To: <sip:+43201@proxy1 Thus, why is there a matching transaction in proxy 1, but not in proxy 2? Why are ACKs not accounting in every case? Thanks Klaus proxy 1 (full debug): ------------------- SIP Request: method: <ACK> uri: <sip:user201@10.10.1.50:8737> version: <SIP/2.0> parse_headers: flags=2 DEBUG: add_param: tag=2b4e3b2d DEBUG:parse_to:end of header reached, state=29 DEBUG: get_hdr_field: <To> [52]; uri=[sip:+43201@proxy1.ienum.labs.nic.at] DEBUG: to body [<sip:+43201@proxy1.ienum.labs.nic.at>] Found param type 232, <branch> = <z9hG4bK-d87543-685643142-1--d87543->; state=6 Found param type 235, <rport> = <n/a>; state=17 end of header reached, state=5 parse_headers: Via found, flags=2 parse_headers: this is the first via After parse_msg... preparing to run routing scripts... DEBUG : sl_filter_ACK: to late to be a local ACK! DEBUG: add_param: tag=f937ab7a DEBUG:parse_to:end of header reached, state=29 xl_printf: final buffer length 186 parse_headers: flags=100 get_hdr_field: cseq <CSeq>: <2> <ACK> DEBUG:maxfwd:is_maxfwd_present: value = 70 parse_headers: flags=200 is_preloaded: No grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5061 matches port 8737 check_self: host != me grep_sock_info - checking if host==us: 10==10 && [10.10.0.41] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 5060 after_loose: Topmost route URI: 'sip:10.10.0.41;lr;ftag=f937ab7a' is me after_loose: URI to be processed: 'sip:10.10.0.41:5062;r2=on;ftag=f937ab7a;lr=on' after_loose: Next URI is a loose router parse_headers: flags=ffffffffffffffff DEBUG: get_hdr_body : content_length=0 found end of header DEBUG: t_newtran: msg id=2 , global msg id=0 , T on entrance=0xffffffff parse_headers: flags=ffffffffffffffff parse_headers: flags=78 t_lookup_request: start searching: hash=6242, isACK=1 parse_headers: flags=38 DEBUG: t_lookup_request: e2e proxy ACK found parse_headers: flags=8 DEBUG: totag for e2e ACK found: 0 DBG: trans=0x406b1940, callback type 4, id 0 entered parse_headers: flags=258 ACC: request acknowledged: method=ACK, i-uri=sip:user201@10.10.1.50:8737, o-uri=sip:user201@10.10.1.50:8737, call_id=9c5e0930eb2b8e5a, from="user101"<sip:user101@proxy1.ienum.labs.nic.at>;tag=f937ab7a, code=200 parse_headers: flags=258 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5061 matches port 8737 check_self: host != me Route ftag and From tag are same DEBUG:tm:t_relay: forwarding ACK statelessly DEBUG: mk_proxy: doing DNS lookup... check_via_address(10.10.1.51, 10.10.1.51, 0) Sending: ACK sip:user201@10.10.1.50:8737 SIP/2.0^M To: <sip:+43201@proxy1 proxy 2 (full debug): ------------------- SIP Request: method: <ACK> uri: <sip:user201@10.10.1.50:8737> version: <SIP/2.0> parse_headers: flags=2 DEBUG: add_param: tag=2b4e3b2d DEBUG:parse_to:end of header reached, state=29 DEBUG: get_hdr_field: <To> [52]; uri=[sip:+43201@proxy1.ienum.labs.nic.at] DEBUG: to body [<sip:+43201@proxy1.ienum.labs.nic.at>] Found param type 232, <branch> = <0>; state=16 end of header reached, state=5 parse_headers: Via found, flags=2 parse_headers: this is the first via After parse_msg... preparing to run routing scripts... DEBUG : sl_filter_ACK: to late to be a local ACK! DEBUG: add_param: tag=f937ab7a DEBUG:parse_to:end of header reached, state=29 xl_printf: final buffer length 186 parse_headers: flags=100 Found param type 232, <branch> = <z9hG4bK-d87543-685643142-1--d87543->; state=6 Found param type 235, <rport> = <6477>; state=16 end of header reached, state=5 parse_headers: Via found, flags=100 parse_headers: this is the second via get_hdr_field: cseq <CSeq>: <2> <ACK> DEBUG:maxfwd:is_maxfwd_present: value = 69 parse_headers: flags=200 is_preloaded: No grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5062 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5062 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5063 matches port 8737 check_self: host != me grep_sock_info - checking if host==us: 10==10 && [10.10.0.41] == [10.10.0.41] grep_sock_info - checking if port 5062 matches port 5062 after_loose: Topmost route URI: 'sip:10.10.0.41:5062;r2=on;ftag=f937ab7a;lr=on' is me grep_sock_info - checking if host==us: 10==10 && [10.10.0.41] == [10.10.0.41] grep_sock_info - checking if port 5063 matches port 5063 after_loose: URI to be processed: 'sip:10.10.0.42:5063;transport=tls;r2=on;ftag=f937ab7a;lr=on' after_loose: Next URI is a loose router DEBUG: has_totag: totag found parse_headers: flags=ffffffffffffffff DEBUG: get_hdr_body : content_length=0 found end of header DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffff parse_headers: flags=ffffffffffffffff parse_headers: flags=78 t_lookup_request: start searching: hash=6242, isACK=1 DEBUG: proceeding to pre-RFC3261 transaction matching DEBUG: t_lookup_request: no transaction found DEBUG:tm:t_relay: forwarding ACK statelessly DEBUG: mk_proxy: doing DNS lookup... parse_headers: flags=1000 check_via_address(10.10.0.41, 10.10.0.41, 0) Sending: ACK sip:user201@10.10.1.50:8737 SIP/2.0^M To: <sip:+43201@proxy1

2 1

[Users] what's about srtp on openser?
by Kenny Yeh 09 Jan '06

09 Jan '06

Bogdan-Andrei Iancu,hello Ask for you again. :) I am succesful on openser for TLS connected by SIPp,but not minisip :( thanks for your more help! Now I step to srtp! Can I use SRTP to build peer to peer communication from openser? if outbound via openser.or do you learn more about srtp of SIP. Best Regard! Kenny from China Shanghai 2006-1-5

3 2

Re: Re: [Users] help for TLS on openser
by Kenny Yeh 09 Jan '06

09 Jan '06

Bogdan-Andrei Iancu,hello Thanks for your help!!!! I try 'netstat -ual' but nothing is list; I try 'netstat -al' list is below: Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State tcp 0 0 debian1:sip-tls *:* LISTEN tcp 0 0 localhost:mysql *:* LISTEN tcp 0 0 *:ftp *:* LISTEN tcp 0 0 localhost:smtp *:* LISTEN tcp6 0 0 *:ssh *:* LISTEN We can see 'debian1' :sip-tls,but I cann't see service port 5061 ,why? or must I connect to server on 5061 port,unlees I cann't see it? Thanks you again. ======= 2006-01-05 02:50:15 您在来信中写道：======= >Hi Kenny, > >Kenny Yeh wrote: > >>users，hello >> >> >> Happly new year! >> >> My problem: >> >> 1.I start openser with TLS support,follow the TLS help file,create a ca/demoCA/ and create cert.pem,private.pem,key.pem.openser is start well,but I cann't list the listen ports by openserctl ps,how I can be sure openser is started by TLS support? >> >> >use "netstat -ual" to check for port 5061 (TLS one). > >> 2.My client is minisip phone that TLS support.but when I start minisip to registar,the error messenge is "exception caught when creating TLS server",I donn't know what's happend. >> >> >ask the minisip guys or their lists. > >regards, >bogdan > >> Please help me ! >> >> >> >>　　　　　　　　致 >>礼！ >> >> >>　　　　　　　　Kenny Yeh >> 上海金叶通讯科技有限公司 >> TEL:8621-6421-6758 ext.311 >>　　　　　　　　kenny(a)artdio.com.tw >>　　　　　　　　　　2006-01-04 >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-05

2 1

[Users] Re:Functionality
by ye jian 09 Jan '06

09 Jan '06

Hi, I think what's you said is a p2p on SIP.that's a good idea for SIP,but who can make it to reality? I cann't ,who? -- 美好的一天,由你和我来创造!

1 0

Re: [Users] snom registration expiry
by Christian Benke 09 Jan '06

09 Jan '06

Hello Klaus! Thanks for your reply, you have triggered some new ideas. of course i should have thought about checking back if any packets are received by the openser host after the line is dead but i thought if it still works with ser, it should work with openser too(and i still have no problems with a second line registered at ser@port5060 while the registration at openser@5061 is lost). ok, the result of the ngrep was that no packets appear on the openser-host when i try to call from the (dead) line. i've made some tracing both on the snom and the ser/openser-host as you suggested. To my surprise, the snom seems to be the device that doesn't behave as expected - it doesn't re-register on the openser-configuration! I have made the attached traces in this way: First i've set the account-details on the snom with 103456(a)sip.de.domain.cc,then rebooted the phone and started the trace on the siprouter-host on port 5060(ser), then changed the configuration to 112233(a)siptest1.domain.cc, rebooted the phone and made the second trace on port 5061(openser). In both cases i didn't touch the phone or make any calls after the reboot to get a pure registration-trace. When i use the account 112233(a)siptest1.domain.cc, the snom registers once and doesn't try to re-register anymore. With the account 103456(a)sip.de.domain.cc, the snom re-registers every 5 minutes, not the expected 10 minutes, but at least it does re-register. When switching the account-details, i didn't do anything but changing the login-details(username, password, registrar-url), the rest of the configuration has not been touched. The phone-firmware is 3.60w. There doesn't seem to be a difference between the siplogs, at least i can't see it. i'm really puzzled right now... 8-| I've attached the siptrace from the ser/openser-host: SER_host_siptrace.txt/OPENSER_host_siptrace.txt the syslog from snom190: SER_snom190_log.txt/OPENSER_snom190_log.txt and the siptrace from snom190: SER_snom_siptrace.txt/OPENSER_snom190_siptrace.txt I have tried this several times now, changing the configuration and rebooting, same results everytime. on SER it's reregistering exactly every 5 minutes, on OPENSER it isn't. I hope all this information is not too much and you have the time to sniff through. thanks! christian p.s.:i've also thought about a error in the srv-records on my nameservers, but they look exactly the same for all sub-domains, the only difference is the port change from 5060 to 5061... On Thu, 05 Jan 2006 10:58:46 +0100 Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > Hi Christian! > The more interesting part would be the SIP signaling between the SNOM and the proxy. If registration is lost, that means that: > - SNOM stops sending REGISTER or > - SNOM still sends REGISTER but they are not accepted by openser Please set exires back to 10min and watch the SIP signaling: > - using "ngrep -t -W byline port 5060" on the proxy > - using the logging feature of the snom phone > Please post this logs. > regards > klaus > Christian Benke wrote: > > Hello! > > > > I've been running my experimental(production in the near future) environment with ser and a mysql-db for some months now but want to switch > > to postgres and openser soon. i'm using a very simple routing as i didn't > > have the time to dig deeper into the ser-routing-language. > > > > i've already adapted my current config-file to openser and the routing works as expected. still, there is one major problem i could not cope with yet: Expiry. > > > > i'm using a snom 190 and had the "proposed expiry" > > set to 10min with ser, without any registration-problems or > > registration-loss. > > with openser with the same settings on the snom i loose my registration > > within a few minutes after phone-reboot, when setting "proposed expiry" to > > 1min on the snom it works without problems endlessly - but setting this > > generally to 1min on all the phones that will register at openser doesn't > > seem to be a good idea and i guess there are other ways. > > > > I've tried to force it with the modparams min_expire 100, max_expire 1800 > > and default_expire 600 > > but the result is still the same, calls are possible after rebooting the > > phone, but after some minutes(below 10min), i loose registration. I had > > similar problems due to a firewall closing the connection, but that occured when i had set a 60min expiry on the snom(with ser), the 10min timeout works fine with the same snom 190 and ser. > > I've searched a lot for > > expiry-problems but only found the modparams above that could be responsible for this. > > Sorry if my explanations sound simple and untechnical, > > but i'm pretty unexperienced with ser/openser yet so i try to explain it > > the way i understand it, simple ;-) > > > > i've attached my config-files, both ser and openser: > > 21X.XXX.XXX.100 is the host running ser(port 5060) and openser(port 5061) > > 21X.XXX.XXX.101 is asterisk1 > > 21X.XXX.XXX.102 is asterisk2(asterisk2.de.domain.cc) > > > > i'm registering my phones at openser with the domain siptest1.domain.cc, the > > second domain sip2.de.domain.cc is a test for a multidomain setup but doesn't work yet(i can't > > use the domain-column provided by the db since i need two different origination domains in asterisk's sip.conf to seperate the calls - maybe > > not the best approach...) > > > > p.s.: > > a second question regarding the multidomain - i can't register with sip2.de.domain.cc - i guess there's something wrong with my regexp > > > > "if (uri=~"^sip:(.+@)?((sip2\.)?(de\.)?domain\.cc)([:;\?].*)?$") {" > > > > someone having mercy and tell me what is wrong with my syntax(or maybe a > > link to a descriptive documentation - ser admin guide was not enough for > > me)? > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users

2 1

RE: [Users] direct manipulation of DB
by Douglas Garstang 09 Jan '06

09 Jan '06

Isn't there a script called 'sc' somewhere that does the same thing as openserctl? I looked at the sc script and was able to determine how it generates those fields. I think they're gen_ha and something else... installed by openser. -----Original Message----- From: unplug [mailto:maillisting@gmail.com] Sent: Sun 1/8/2006 7:50 PM To: Norman Brandinger Cc: users(a)openser.org Subject: Re: [Users] direct manipulation of DB There are 2 parts. One is server and one is client. How to setup the server as openser has already a fifo interfece? On 12/23/05, Norman Brandinger <norm(a)goes.com> wrote: > I posted a simple FIFO Client that can be used to test sending commands > and receiving responses from OpenSER. > > The link is here: http://openser.org/dokuwiki/doku.php?id=utilities > > Regards, > Norm > > p.s. Comments or suggestions for improvements are welcome. > > p.s.s. I believe that the OpenSER roadmap has an API project that will > depreciate the FIFO interface. Perhaps one of the developers can > elaborate more on this. > > Klaus Darilion wrote: > > > unplug wrote: > > > >> I am using openser with mysql as db. There is a tool call openserctl > >> to manage the data in the openser. I wonder whether I can direct > >> manipulate the db directly without using openserctl. I have tried to > >> insert a record in the aliases table directly using mysql client but > >> it doesn't work. Can I conclude direct manipulate the DB is > >> impossible? > > > > > > It depends on the table! There are some tables which will be cached > > inside openser, e.g: > > location > > aliases > > domain > > gw (lcr module) > > > > If you make changes to location/aliases you have to: > > - restart openser to reload the tables or > > - use openserctl (inserts via FIFO to openser which wirtes back to > > the db or > > - in case of aliases use the aliases_DB module > > > > for the domain and gw table, you can use the respectife FIfo commands > > to reload the tables. > > > > regards > > klaus > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

RE: [Users] Re: OpenSER and transport selection (SRV and NAPTR)
by Douglas Garstang 09 Jan '06

09 Jan '06

Oh yes! That would be AMAZING if that functionality could be added! I think I'm gonna poo my pants! -----Original Message----- From: Greg Fausak [mailto:lgfausak@gmail.com] Sent: Fri 1/6/2006 11:06 AM To: users(a)openser.org Cc: Subject: [Users] Re: OpenSER and transport selection (SRV and NAPTR) Howdy guys, I completely missed this thread. How is this going? I just read the roadmap which indicates NAPTR lookup is on the list. I've been maintaining NAPTR and SRV records looking forward to the day when t_relay() forwards using NAPTR (and SRV). Does that mean that the t_relay() will maintain the state, slogging through the NAPTR/SRV records appropriately, without me having to do additional logic??? Specifically, I'm keen on the SRV serial forking. -g Hi Bogdan, > I meant t_reply() will keep its behaviour as looking into URI for the > destination - but it will incorporate the NAPTR lookup. Great - this answers my question. > to response also to on earlier question, regarding the timing > - I say 3 months are more than enough ;). Excellent. Many thanks for your help (and patience ;) --Joachim > Joachim Fabini wrote: > > >Hi Bogdan, > > > > > > > >>indeed, there was a misunderstanding :) .t_relay() with no > >>param will be kept with the current functionality. > >>Or you suggest to be able to specify only a different proto > >>or port from the RURI? > >> > >> > > > >I did not yet find the primitive which is supposed to > >statefully relay and do the following: > >1. NAPTR query on the transport to use PLUS > >2. _exactly_ what t_relay() does now for the > > transport returned by the NAPTR query. > > > >You say that t_relay() is kept with the current functionality. > >Does this mean no NAPTR, or will t_relay() be extended to use > >NAPTR before SRV/A query? If the latter then everything is OK. > > > >Otherwise we have the alternatives: > > > >t_relay() -> Stateful relaying according to destination > > address using the incoming transport (no NAPTR). > >t_relay_to() -> Stateful relaying to a specific host (you said > > that <host> is mandatory here) using NAPTR to > > determine the transport. > > > > > >To summarize: > >My point is that none of these two primitives covers the case > >when the message is to be relayed to the next hop based only > >on the destination address _and_ using the transport selected > >by the destination proxy (determined via NAPTR query). > > > >Except if either a) t_relay() does NAPTR or b) the host > >parameter in t_relay_to() is optional. > > > >--Joachim > > > > > > > > > -- Greg Fausak greg(a)thursday.com _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

Re: [Users] deactivate an account
by Helge Waastad 09 Jan '06

09 Jan '06

Hi, In the openserctl script there are acl options static configured (free-pstn etc). Just add inactive on the same line, and it should work br hw There is an error when I issue acl grant openserctl acl grant myname(a)mydomain.com inactive Invalid privilege: inactive ignored -- Mvh/Br Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 1

[Users] direct manipulation of DB
by unplug 09 Jan '06

09 Jan '06

I am using openser with mysql as db. There is a tool call openserctl to manage the data in the openser. I wonder whether I can direct manipulate the db directly without using openserctl. I have tried to insert a record in the aliases table directly using mysql client but it doesn't work. Can I conclude direct manipulate the DB is impossible?

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[Users] Re: Users Digest, Vol 8, Issue 14
by Kenny Yeh 09 Jan '06

09 Jan '06

Hi,Philipp Thanks for your help !!!!!! You show me what I need .thanks!!! ======= 2006-01-07 03:15:05 ��д��======= >Send Users mailing list submissions to > users(a)openser.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://openser.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to > users-request(a)openser.org > >You can reach the person managing the list at > users-owner(a)openser.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > > 1. uac_replace_from function (unplug) > 2. Re: NAT+uac_replace_from (Daniel-Constantin Mierla) > 3. Re: uac_replace_from function (Daniel-Constantin Mierla) > 4. Re: OpenSER and Solaris (Daniel-Constantin Mierla) > 5. Re: OpenSER and transport selection (SRV and NAPTR) (Greg Fausak) > 6. Re: help: what's tls_ca tls_privat-key tls_calist > (Alexander Philipp Lintenhofer) > 7. contact field port problem when behind nats (Jack Wei) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Fri, 6 Jan 2006 19:08:34 +0800 >From: unplug <maillisting(a)gmail.com> >Subject: [Users] uac_replace_from function >To: users(a)openser.org >Message-ID: > <6fbb529e0601060308p5a0eab32wafce0fcc5f04b286(a)mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 > >I am using uac_replace_from function as below. > ># replace only display and do not touch uri >uac_replace_from("35418474",""); > >After running uac_replace_from function, I get the following message. > >U 203.193.4.242:5060 -> 203.193.4.226:5060 >SIP/2.0 183 Session Progress. >Via: SIP/2.0/UDP >203.193.4.226;branch=z9hG4bKaed7.46cc4101.0,SIP/2.0/UDP >10.0.0.52:5060;rport=5060;received=210.184.4.31;branch=z9hG4bKHNHpldVsKca6IYUS. >From: "35418474"<sip:882754853589@ow01.owtel.com>;tag=sBl2gLH6O1myyqL5. >To: "92588819" <sip:92588819@ow01.owtel.com>;tag=47C61DF4-26CD. >Date: Fri, 06 Jan 2006 10:41:17 GMT. >Call-ID: yTCQvS56ZCZXCHgt(a)10.0.0.52. >Server: Cisco-SIPGateway/IOS-12.x. > >In the normal message, sip message, there is a space between display >name and uri. >ie. From: "35418474" <sip:882754853589@ow01.owtel.com> >However, after I apply uac_replace_from function, there is a space >there. How can I add a space such that it looks like a normal sip >message? Does the space will affect the caller display? > > > >------------------------------ > >Message: 2 >Date: Fri, 06 Jan 2006 13:28:52 +0200 >From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> >Subject: Re: [Users] NAT+uac_replace_from >To: unplug <maillisting(a)gmail.com> >Cc: users(a)openser.org >Message-ID: <43BE5474.4010400(a)voice-system.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Send me the network traces for this case to see what is wrong. The uac >module has a mode when it should do automatic replacement in subsequent >requests. > >Daniel > > >On 01/06/06 04:37, unplug wrote: >> Thanks for your reply. According to your reply, I add the following >> code for the PRACK. >> if (method=="PRACK") { >> if (avp_db_load("$from/uri","s:alias")) { >> xlog("L_INFO","sip408: have alias - [$avp(s:alias)]\n"); >> uac_replace_from("anonymous","sip:$avp(s:alias)@$si"); >> } else { >> xlog("L_INFO","sip411: no alias\n"); >> }; >> } >> >> The result is the same as before. It seems there is a conflict in the >> from header and it causes the 481 error. To be more precise, below is >> the protocol flow. >> >> UA1 server1 server2 >> fromtag:taga -----from username:9000 INVITE-----> >> fromtag:taga <----from username:9000 100 trying--- >> after uac_replace_from("1234","sip:1234@$si") >> fromtag:taga ------from username:1234 INVITE ----> >> (callee rings but caller drops) >> fromtag:taga <-----from username:1234 100 trying---- >> (callee still rings until answers the call) >> fromtag:taga <----from username:1234 session progress 183--- >> fromtag:taga <-----from username:9000 session prgress 183---- >> fromtag:taga -----from username:9000 PRACK-----------> >> fromtag:taga ----------from username:9000 PRACK---> >> fromtag:taga <----from username:9000 481 call leg ----- >> fromtag:taga <----from username:9000 481 call leg ----- >> >> >> On 1/5/06, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> >>> Hello, >>> >>> "SIP/2.0 481 Call Leg/Transaction Does Not Exist" is for PRACK, because you do not change the From header for it. You have to do the same translation for all requests within the dialog. See the documentation of uac module: . >>> >>> http://openser.org/docs/modules/1.1.x/uac.html >>> >>> Cheers, >>> Daniel >>> >>> >>> On 01/04/06 12:21, unplug wrote: >>> >>>> Actually, I am replacing the username of the uri with the alias that >>>> stored in the database. In my configuration file, it is using >>>> mediaproxy for NAT function (features-callfwd.5.0.cfg from getting >>>> started). I also add the following codes in the very first of the >>>> route routine for alias replacing purpose. >>>> >>>> route { >>>> ... >>>> if (!has_totag() && method=="INVITE") { >>>> if (avp_db_load("$from/uri","s:alias")) { >>>> xlog("L_INFO","sip408: have alias - [$avp(s:alias)]\n"); >>>> uac_replace_from("anonymous","sip:$avp(s:alias)@$si"); >>>> } else { >>>> xlog("L_INFO","sip411: no alias\n"); >>>> }; >>>> }; >>>> ... >>>> } >>>> >>>> When I make a call from a phone to the PSTN phone, the caller drops >>>> the ring when the callee rings. Callee hangs up and callee rings >>>> again few seconds after. You can find an error message "SIP/2.0 481 >>>> Call Leg/Transaction Does Not Exist" in line 180. I wonder if there >>>> is any thing wrong with the above code or my concept is wrong. Please >>>> help. Below is the sip message. >>>> [...] >>>> >>>> >> >> > > > >------------------------------ > >Message: 3 >Date: Fri, 06 Jan 2006 13:33:32 +0200 >From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> >Subject: Re: [Users] uac_replace_from function >To: unplug <maillisting(a)gmail.com> >Cc: users(a)openser.org >Message-ID: <43BE558C.70404(a)voice-system.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >It might be a bug in the module, please send the original SIP reply as >well, to be able to trace the problem quicker. > >Daniel > > >On 01/06/06 13:08, unplug wrote: >> I am using uac_replace_from function as below. >> >> # replace only display and do not touch uri >> uac_replace_from("35418474",""); >> >> After running uac_replace_from function, I get the following message. >> >> U 203.193.4.242:5060 -> 203.193.4.226:5060 >> SIP/2.0 183 Session Progress. >> Via: SIP/2.0/UDP >> 203.193.4.226;branch=z9hG4bKaed7.46cc4101.0,SIP/2.0/UDP >> 10.0.0.52:5060;rport=5060;received=210.184.4.31;branch=z9hG4bKHNHpldVsKca6IYUS. >> From: "35418474"<sip:882754853589@ow01.owtel.com>;tag=sBl2gLH6O1myyqL5. >> To: "92588819" <sip:92588819@ow01.owtel.com>;tag=47C61DF4-26CD. >> Date: Fri, 06 Jan 2006 10:41:17 GMT. >> Call-ID: yTCQvS56ZCZXCHgt(a)10.0.0.52. >> Server: Cisco-SIPGateway/IOS-12.x. >> >> In the normal message, sip message, there is a space between display >> name and uri. >> ie. From: "35418474" <sip:882754853589@ow01.owtel.com> >> However, after I apply uac_replace_from function, there is a space >> there. How can I add a space such that it looks like a normal sip >> message? Does the space will affect the caller display? >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > >------------------------------ > >Message: 4 >Date: Fri, 06 Jan 2006 19:41:19 +0200 >From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> >Subject: Re: [Users] OpenSER and Solaris >To: martin(a)campus-merseburg.de >Cc: users(a)openser.org >Message-ID: <43BEABBF.3090302(a)voice-system.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >thanks for reporting, I have renamed the internal structure meminfo to >mem_info to avoid this conflict. Please take the latest 1.0.0 from cvs >and let me know if it works. > >Here are the commands to get the stable version from cvs: > ># export CVSROOT=:pserver:anonymous@cvs.sourceforge.net:/cvsroot/openser ># cvs login - for password simply press the Enter key. ># cvs -z3 checkout -r rel_1_0_0 sip-server > > >Cheers, >Daniel > > >On 01/06/06 12:45, martin(a)campus-merseburg.de wrote: >> Hi, >> >> does anyone use OpenSER under Solaris? >> >> I get this error when compiling: >> >> gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc >> -DNAME='"openser"' -DVERSION='"1.0.0"' -DARCH='"sparc64"' >> -DOS='"solaris"' -DCOMPILER='"gcc 3.4"' -D__CPU_sparc64 >> -D__OS_solaris -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC >> -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP >> -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK >> -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME >> -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -c action.c -o >> action.o >> In file included from mem/f_malloc.h:40, >> from mem/mem.h:54, >> from action.c:50: >> mem/meminfo.h:33: error: redefinition of `struct meminfo' >> gmake: *** [action.o] Error 1 >> >> any idea for fixing this? >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > >------------------------------ > >Message: 5 >Date: Fri, 6 Jan 2006 12:06:42 -0600 >From: Greg Fausak <lgfausak(a)gmail.com> >Subject: [Users] Re: OpenSER and transport selection (SRV and NAPTR) >To: users(a)openser.org >Message-ID: > <8c7b802f0601061006p373d8949we5e48c9953eb8dc9(a)mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 > >Howdy guys, > >I completely missed this thread. >How is this going? > >I just read the roadmap which indicates NAPTR lookup >is on the list. I've been maintaining NAPTR and SRV records >looking forward to the day when t_relay() forwards using NAPTR (and >SRV). Does that mean that the t_relay() will maintain the state, slogging >through the NAPTR/SRV records appropriately, without me having >to do additional logic??? Specifically, I'm keen on the SRV serial forking. > >-g > > > > >Hi Bogdan, > >> I meant t_reply() will keep its behaviour as looking into URI for the >> destination - but it will incorporate the NAPTR lookup. > >Great - this answers my question. > > >> to response also to on earlier question, regarding the timing >> - I say 3 months are more than enough ;). > >Excellent. > >Many thanks for your help (and patience ;) >--Joachim > > > >> Joachim Fabini wrote: >> >> >Hi Bogdan, >> > >> > >> > >> >>indeed, there was a misunderstanding :) .t_relay() with no >> >>param will be kept with the current functionality. >> >>Or you suggest to be able to specify only a different proto >> >>or port from the RURI? >> >> >> >> >> > >> >I did not yet find the primitive which is supposed to >> >statefully relay and do the following: >> >1. NAPTR query on the transport to use PLUS >> >2. _exactly_ what t_relay() does now for the >> > transport returned by the NAPTR query. >> > >> >You say that t_relay() is kept with the current functionality. >> >Does this mean no NAPTR, or will t_relay() be extended to use >> >NAPTR before SRV/A query? If the latter then everything is OK. >> > >> >Otherwise we have the alternatives: >> > >> >t_relay() -> Stateful relaying according to destination >> > address using the incoming transport (no NAPTR). >> >t_relay_to() -> Stateful relaying to a specific host (you said >> > that <host> is mandatory here) using NAPTR to >> > determine the transport. >> > >> > >> >To summarize: >> >My point is that none of these two primitives covers the case >> >when the message is to be relayed to the next hop based only >> >on the destination address _and_ using the transport selected >> >by the destination proxy (determined via NAPTR query). >> > >> >Except if either a) t_relay() does NAPTR or b) the host >> >parameter in t_relay_to() is optional. >> > >> >--Joachim >> > >> > >> > >> > >> >-- >Greg Fausak >greg(a)thursday.com > > > >------------------------------ > >Message: 6 >Date: Fri, 06 Jan 2006 19:47:10 +0100 >From: Alexander Philipp Lintenhofer <lintenhofer(a)aon.at> >Subject: Re: [Users] help: what's tls_ca tls_privat-key tls_calist >To: users(a)openser.org >Message-ID: <20060106194710.x2agnokfltq8ok4s(a)liho.dyndns.org> >Content-Type: text/plain; charset=UTF-8; format="flowed" > >Hi Kenny, > >It would be a big deal first to acquire knowledge about the theoretical >background before installing this service. > >So please read following (and related) sites: >http://www.tech-invite.com/Ti-sip-security.html >http://www.tech-invite.com/Ti-SSL.html > >The definitive documentation about OpenSER/TLS >(http://openser.org/docs/tls.html) is very good and therefore easy to >understand in succession! > >regards, >Philipp > >> Hello all, >> >> I have tested tls with openser by SIPp,but I donn't understant >> the flows of TLS ctreated,who can show me the flow about TLS >> handshake. >> >> When I test tls by SIPp,I put the ca,private-key to it's >> DIR,that's fils created by openssl ,and add these files path to cfg >> of openser.the same files used.I want to know what's mean is tls_ca >> tls_privatekey tls_calist,and how there files work. Client need >> what's ca and private-key for creating tls connection? Is it the same >> with openssl? to create ssl connection? >> >> >> >> >> >> ======= 2006-01-05 19:01:27 您在来信中写道：======= >> >>> Send Users mailing list submissions to >>> users(a)openser.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> or, via email, send a message with subject or body 'help' to >>> users-request(a)openser.org >>> >>> You can reach the person managing the list at >>> users-owner(a)openser.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of Users digest..." >>> >>> >>> Today's Topics: >>> >>> 1. Re: deactivate an account (Helge Waastad) >>> >>> >>> ---------------------------------------------------------------------- >>> >>> Message: 1 >>> Date: Thu, 05 Jan 2006 11:23:30 +0100 >>> From: Helge Waastad <helge(a)smartnet.no> >>> Subject: Re: [Users] deactivate an account >>> To: maillisting(a)gmail.com >>> Cc: users(a)openser.org >>> Message-ID: <1136456610.11157.10.camel(a)localhost.localdomain> >>> Content-Type: text/plain >>> >>> Hi, >>> In the openserctl script there are acl options static configured >>> (free-pstn etc). >>> Just add inactive on the same line, and it should work >>> >>> br hw >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> There is an error when I issue acl grant >>> openserctl acl grant myname(a)mydomain.com inactive >>> Invalid privilege: inactive ignored >>> >>> >>> >>> -- >>> Mvh/Br >>> Helge Waastad >>> Senior Engineer >>> Systemavdelingen >>> Smartnet >>> >>> >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> End of Users Digest, Vol 8, Issue 11 >>> ************************************ >> >> = = = = = = = = = = = = = = = = = = = = >> >> >> 　　　　　　　　�? >> 礼！ >> >> >> 　　　　　　　　Kenny Yeh >> 　　　　　　　　kenny(a)artdio.com.tw >> 　　　　　　　　　　2006-01-06 >> >> > > > > > > >------------------------------ > >Message: 7 >Date: Fri, 6 Jan 2006 11:14:14 -0800 (PST) >From: Jack Wei <cowlemon(a)yahoo.com> >Subject: [Users] contact field port problem when behind nats >To: users(a)openser.org >Message-ID: <20060106191415.56473.qmail(a)web50008.mail.yahoo.com> >Content-Type: text/plain; charset=iso-8859-1 > >Hi, > >I'm having a problem when an UAC is behind NAT and has a public port other than >5060. The public port isn't present in the contact field so when an ACK for an >INVITE is sent to the UAC, it's directed to the default port of 5060. I'm >currently using v1.0.0 w/ the NATHELPER module. The UAC is SJphone. Any help >would be greatly appreciated. > > > > > >NAT part of my openser.cfg : > >if (nat_uac_test("19")) >{ > if ( method == "REGISTER" ) > { > fix_nated_register(); > force_rport(); > } > else > { > force_rport(); > fix_nated_contact(); > > if (method == "INVITE") > { > fix_nated_sdp("1"); > }; > }; > > append_hf( "P-hint: NAT traversal\r\n" ); >} > > > > >the sip proxy is voip.somewhere.com and 1.1.1.1 >the pstn gateway is 2.2.2.2 > >the caller is from pstn and the callee is the uac. > > >Here's a log of SIP msgs: > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >INVITE sip:2139435286@voip.somewhere.com:5060;user=phone SIP/2.0. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >Remote-Party-Id: ><sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 70. >m: <sip:2139961998@2.2.2.2:5060;user=phone>. >k: replaces. >c: application/sdp. >Accept: application/sdp. >Accept-Encoding: . >Accept-Language: en. >User-Agent: Lucent-Universal-Gateway. >l: 244. >. >v=0. >o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. >s=Session SDP. >c=IN IP4 2.2.2.2. >t=0 0. >m=audio 41788 RTP/AVP 0 101. >a=silenceSupp:off. >a=ecan:b on g168. >a=ptime:20. >a=rtpmap:101 telephone-event/8000. >a=rtpmap:0 PCMU/8000. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >SIP/2.0 100 trying -- your call is important to us. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >v: SIP/2.0/UDP 2.2.2.2:5060. >Server: OpenSer (1.0.0-tls (i386/linux)). >Content-Length: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:32883 >INVITE sip:2139435286@64.77.236.227 SIP/2.0. >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >Remote-Party-Id: ><sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK8325.35c96c63.0. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 69. >m: <sip:2139961998@2.2.2.2:5060;user=phone>. >k: replaces. >c: application/sdp. >Accept: application/sdp. >Accept-Encoding: . >Accept-Language: en. >User-Agent: Lucent-Universal-Gateway. >l: 244. >P-hint: VOIP OUTBOUND. >. >v=0. >o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. >s=Session SDP. >c=IN IP4 2.2.2.2. >t=0 0. >m=audio 41788 RTP/AVP 0 101. >a=silenceSupp:off. >a=ecan:b on g168. >a=ptime:20. >a=rtpmap:101 telephone-event/8000. >a=rtpmap:0 PCMU/8000. > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >SIP/2.0 100 Trying. >v: SIP/2.0/UDP >voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP >2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: ""<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 0. >Server: SJphone/1.61.306c (SJ Labs). >. > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP >voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP >2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP 2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on >SIP/2.0. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 70. >Route: <sip:2139435286@64.77.236.227>. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:5060 >ACK sip:2139435286@64.77.236.227 SIP/2.0. >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >Via: SIP/2.0/UDP voip.somewhere.com;branch=0. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 69. >User-Agent: Lucent-Universal-Gateway. >l: 0. >P-hint: rr-enforced. >. > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP >voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP >2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP 2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on >SIP/2.0. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 70. >Route: <sip:2139435286@64.77.236.227>. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:5060 >ACK sip:2139435286@64.77.236.227 SIP/2.0. >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >Via: SIP/2.0/UDP voip.somewhere.com;branch=0. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 69. >User-Agent: Lucent-Universal-Gateway. >l: 0. >P-hint: rr-enforced. >. > > > > > > >................<REPEAT OKs and ACKs>........................ > > > > > > > > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. >v: SIP/2.0/UDP >192.168.0.100;rport;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >f: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Max-Forwards: 70. >User-Agent: SJphone/1.61.306c (SJ Labs). >l: 0. >Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. >Record-Route: <sip:2139961998@voip.somewhere.com;ftag=71451719939;lr=on>. >Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. >v: SIP/2.0/UDP >192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >f: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Max-Forwards: 69. >User-Agent: SJphone/1.61.306c (SJ Labs). >l: 0. >P-hint: NAT traversal. >P-hint: rr-enforced. >. > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. >v: SIP/2.0/UDP >192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >f: "Jack Wei" ><sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Server: Lucent-Universal-Gateway. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:32883 >SIP/2.0 200 OK. >v: SIP/2.0/UDP >192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >f: "Jack Wei" ><sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Server: Lucent-Universal-Gateway. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > > >Jack Wei > > > >__________________________________________ >Yahoo! DSL ?Something to write home about. >Just $16.99/mo. or less. >dsl.yahoo.com > > > > >------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >End of Users Digest, Vol 8, Issue 14 >************************************ = = = = = = = = = = = = = = = = = = = = �� Kenny Yeh ��kenny(a)artdio.com.tw ��2006-01-09

1 0

[Users] contact field port problem when behind nats
by Jack Wei 07 Jan '06

07 Jan '06

Hi, I'm having a problem when an UAC is behind NAT and has a public port other than 5060. The public port isn't present in the contact field so when an ACK for an INVITE is sent to the UAC, it's directed to the default port of 5060. I'm currently using v1.0.0 w/ the NATHELPER module. The UAC is SJphone. Any help would be greatly appreciated. NAT part of my openser.cfg : if (nat_uac_test("19")) { if ( method == "REGISTER" ) { fix_nated_register(); force_rport(); } else { force_rport(); fix_nated_contact(); if (method == "INVITE") { fix_nated_sdp("1"); }; }; append_hf( "P-hint: NAT traversal\r\n" ); } the sip proxy is voip.somewhere.com and 1.1.1.1 the pstn gateway is 2.2.2.2 the caller is from pstn and the callee is the uac. Here's a log of SIP msgs: # U 2.2.2.2:5060 -> 1.1.1.1:5060 INVITE sip:2139435286@voip.somewhere.com:5060;user=phone SIP/2.0. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. Remote-Party-Id: <sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 70. m: <sip:2139961998@2.2.2.2:5060;user=phone>. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: Lucent-Universal-Gateway. l: 244. . v=0. o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. s=Session SDP. c=IN IP4 2.2.2.2. t=0 0. m=audio 41788 RTP/AVP 0 101. a=silenceSupp:off. a=ecan:b on g168. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. # U 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 100 trying -- your call is important to us. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. v: SIP/2.0/UDP 2.2.2.2:5060. Server: OpenSer (1.0.0-tls (i386/linux)). Content-Length: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:32883 INVITE sip:2139435286@64.77.236.227 SIP/2.0. Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. Remote-Party-Id: <sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK8325.35c96c63.0. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 69. m: <sip:2139961998@2.2.2.2:5060;user=phone>. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: Lucent-Universal-Gateway. l: 244. P-hint: VOIP OUTBOUND. . v=0. o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. s=Session SDP. c=IN IP4 2.2.2.2. t=0 0. m=audio 41788 RTP/AVP 0 101. a=silenceSupp:off. a=ecan:b on g168. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. # U 64.77.236.227:32883 -> 1.1.1.1:5060 SIP/2.0 100 Trying. v: SIP/2.0/UDP voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: ""<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 0. Server: SJphone/1.61.306c (SJ Labs). . # U 64.77.236.227:32883 -> 1.1.1.1:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on SIP/2.0. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 70. Route: <sip:2139435286@64.77.236.227>. User-Agent: Lucent-Universal-Gateway. l: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:5060 ACK sip:2139435286@64.77.236.227 SIP/2.0. Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. Via: SIP/2.0/UDP voip.somewhere.com;branch=0. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 69. User-Agent: Lucent-Universal-Gateway. l: 0. P-hint: rr-enforced. . # U 64.77.236.227:32883 -> 1.1.1.1:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on SIP/2.0. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 70. Route: <sip:2139435286@64.77.236.227>. User-Agent: Lucent-Universal-Gateway. l: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:5060 ACK sip:2139435286@64.77.236.227 SIP/2.0. Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. Via: SIP/2.0/UDP voip.somewhere.com;branch=0. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 69. User-Agent: Lucent-Universal-Gateway. l: 0. P-hint: rr-enforced. . ................<REPEAT OKs and ACKs>........................ # U 64.77.236.227:32883 -> 1.1.1.1:5060 BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. v: SIP/2.0/UDP 192.168.0.100;rport;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. f: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Max-Forwards: 70. User-Agent: SJphone/1.61.306c (SJ Labs). l: 0. Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . # U 1.1.1.1:5060 -> 2.2.2.2:5060 BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. Record-Route: <sip:2139961998@voip.somewhere.com;ftag=71451719939;lr=on>. Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. v: SIP/2.0/UDP 192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. f: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Max-Forwards: 69. User-Agent: SJphone/1.61.306c (SJ Labs). l: 0. P-hint: NAT traversal. P-hint: rr-enforced. . # U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. v: SIP/2.0/UDP 192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. f: "Jack Wei" <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Server: Lucent-Universal-Gateway. User-Agent: Lucent-Universal-Gateway. l: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:32883 SIP/2.0 200 OK. v: SIP/2.0/UDP 192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. f: "Jack Wei" <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Server: Lucent-Universal-Gateway. User-Agent: Lucent-Universal-Gateway. l: 0. . Jack Wei __________________________________________ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com

2 1

[Users] help: what's tls_ca tls_privat-key tls_calist
by Kenny Yeh 06 Jan '06

06 Jan '06

Hello all, I have tested tls with openser by SIPp,but I donn't understant the flows of TLS ctreated,who can show me the flow about TLS handshake. When I test tls by SIPp,I put the ca,private-key to it's DIR,that's fils created by openssl ,and add these files path to cfg of openser.the same files used.I want to know what's mean is tls_ca tls_privatekey tls_calist,and how there files work. Client need what's ca and private-key for creating tls connection? Is it the same with openssl? to create ssl connection? ======= 2006-01-05 19:01:27 您在来信中写道：======= >Send Users mailing list submissions to > users(a)openser.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://openser.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to > users-request(a)openser.org > >You can reach the person managing the list at > users-owner(a)openser.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > > 1. Re: deactivate an account (Helge Waastad) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 05 Jan 2006 11:23:30 +0100 >From: Helge Waastad <helge(a)smartnet.no> >Subject: Re: [Users] deactivate an account >To: maillisting(a)gmail.com >Cc: users(a)openser.org >Message-ID: <1136456610.11157.10.camel(a)localhost.localdomain> >Content-Type: text/plain > >Hi, >In the openserctl script there are acl options static configured >(free-pstn etc). >Just add inactive on the same line, and it should work > >br hw > > > > > > > > > > > >There is an error when I issue acl grant >openserctl acl grant myname(a)mydomain.com inactive >Invalid privilege: inactive ignored > > > >-- >Mvh/Br >Helge Waastad >Senior Engineer >Systemavdelingen >Smartnet > > > >------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >End of Users Digest, Vol 8, Issue 11 >************************************ = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-06

2 1

[Users] OpenSER and Solaris
by martin＠campus-merseburg.de 06 Jan '06

06 Jan '06

Hi, does anyone use OpenSER under Solaris? I get this error when compiling: gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.0.0"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4"' -D__CPU_sparc64 -D__OS_solaris -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -c action.c -o action.o In file included from mem/f_malloc.h:40, from mem/mem.h:54, from action.c:50: mem/meminfo.h:33: error: redefinition of `struct meminfo' gmake: *** [action.o] Error 1 any idea for fixing this?

2 1

[Users] 500 command 'domain_dump' not available
by Bobber Cheng 06 Jan '06

06 Jan '06

I use openser-1.0.0. #/usr/local/sbin/openserctl domain show 500 command 'domain_dump' not available i load domain.so in my openser.cf like loadmodule "/usr/local/lib/openser/modules/domain.so"

3 4

Re: Re: [Users] what's about srtp on openser?
by Kenny Yeh 05 Jan '06

05 Jan '06

Klaus Darilion,hi You mean is if I add SRTP to re-place RTP of client,Client can create srtp to communication?? So I just re-build my client code is ok?it's changed by udp or SIP lays or both? ======= 2006-01-05 17:53:26 您在来信中写道：======= >srtp is independent from SIP. There is no need to support srtp on the >SIP proxy. If you use mediaproxy or rtpproxy (rtp relay), it should work >too as they do not care about the payload, just forward the packets. > >regards >klaus > >Kenny Yeh wrote: >> Bogdan-Andrei Iancu,hello >> >> Ask for you again. :) >> >> I am succesful on openser for TLS connected by SIPp,but not minisip :( >> thanks for your more help! >> Now I step to srtp! Can I use SRTP to build peer to peer communication from openser? if outbound via openser.or do you learn more about srtp of SIP. >> >> >> Best Regard! >> >> Kenny from China Shanghai >> 2006-1-5 >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-05

2 1

[Users] deactivate an account
by unplug 05 Jan '06

05 Jan '06

Hi , Does anyone know how can I deactivate an account? Does openser already implement that function to deactivate an account, how?

5 7

[Users] openser crashed
by Jayesh Nambiar 05 Jan '06

05 Jan '06

hi all, I am using openser 0.9.5. I recently configured openser supporting gateway chaining with two failure routes. I tested it and it was working fine, and when I put it into actual production environment which processes close to 200 calls per second in peak hrs, openser crashed after an hours time. I think there was a memory leak. The error message in syslog at that time showed: ERROR: t_should_relay_response: status rewrite by UAS: stored: 487, received: 100 my snippet of cfg file is below: modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_timer", 3) modparam("tm", "fr_inv_timer_avp", "3") if(method=="INVITE") { setflag(1); if (uri=~"^sip:00[0-9]*@") { strip(2); rewritehost("216.XXX.XXX.XX"); t_on_failure("1"); t_on_reply("2"); t_relay(); break; }; failure_route[1] { rewritehost("217.XXX.XXX.XX"); append_branch(); t_on_failure("2"); t_relay(); } failure_route[2] { rewritehost("218.XXX.XXX.XX"); append_branch(); t_relay(); } onreply_route[2] { if (status=~"18[0-9]") { t_on_failure("0"); }; } I had ser fr_timer to 3 secs so that if there is no response within 3 secs the other route is tried. Please let me know what logical error was there in my cfg which made openser crash. Is it something related with the timer? thanks in advance. jayesh Send instant messages to your online friends http://in.messenger.yahoo.com

2 1

[Users] snom registration expiry
by Christian Benke 05 Jan '06

05 Jan '06

Hello! I've been running my experimental(production in the near future) environment with ser and a mysql-db for some months now but want to switch to postgres and openser soon. i'm using a very simple routing as i didn't have the time to dig deeper into the ser-routing-language. i've already adapted my current config-file to openser and the routing works as expected. still, there is one major problem i could not cope with yet: Expiry. i'm using a snom 190 and had the "proposed expiry" set to 10min with ser, without any registration-problems or registration-loss. with openser with the same settings on the snom i loose my registration within a few minutes after phone-reboot, when setting "proposed expiry" to 1min on the snom it works without problems endlessly - but setting this generally to 1min on all the phones that will register at openser doesn't seem to be a good idea and i guess there are other ways. I've tried to force it with the modparams min_expire 100, max_expire 1800 and default_expire 600 but the result is still the same, calls are possible after rebooting the phone, but after some minutes(below 10min), i loose registration. I had similar problems due to a firewall closing the connection, but that occured when i had set a 60min expiry on the snom(with ser), the 10min timeout works fine with the same snom 190 and ser. I've searched a lot for expiry-problems but only found the modparams above that could be responsible for this. Sorry if my explanations sound simple and untechnical, but i'm pretty unexperienced with ser/openser yet so i try to explain it the way i understand it, simple ;-) i've attached my config-files, both ser and openser: 21X.XXX.XXX.100 is the host running ser(port 5060) and openser(port 5061) 21X.XXX.XXX.101 is asterisk1 21X.XXX.XXX.102 is asterisk2(asterisk2.de.domain.cc) i'm registering my phones at openser with the domain siptest1.domain.cc, the second domain sip2.de.domain.cc is a test for a multidomain setup but doesn't work yet(i can't use the domain-column provided by the db since i need two different origination domains in asterisk's sip.conf to seperate the calls - maybe not the best approach...) p.s.: a second question regarding the multidomain - i can't register with sip2.de.domain.cc - i guess there's something wrong with my regexp "if (uri=~"^sip:(.+@)?((sip2\.)?(de\.)?domain\.cc)([:;\?].*)?$") {" someone having mercy and tell me what is wrong with my syntax(or maybe a link to a descriptive documentation - ser admin guide was not enough for me)?

2 1

[Users] lookup function for From URI
by unplug 05 Jan '06

05 Jan '06

As I know, we can use lookup function for lookup an alias of an RURI. It is for inbound call. How about the outbound call? Say, I have an account (881234) with an alias (24568849). Someone can call the alias to reach the account (881234) by using lookup function. How can I make outbound call such that the caller number will be replaced by the alias? As I know we can use uac_replace_from(display,uri) to replace the from URI. But I have no idea how to retrieve an alias from the >From URI. Any clue?

2 4

[Users] help for TLS on openser
by Kenny Yeh 04 Jan '06

04 Jan '06

users，hello Happly new year! My problem: 1.I start openser with TLS support,follow the TLS help file,create a ca/demoCA/ and create cert.pem,private.pem,key.pem.openser is start well,but I cann't list the listen ports by openserctl ps,how I can be sure openser is started by TLS support? 2.My client is minisip phone that TLS support.but when I start minisip to registar,the error messenge is "exception caught when creating TLS server",I donn't know what's happend. Please help me ! 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 上海金叶通讯科技有限公司 TEL:8621-6421-6758 ext.311 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-04

2 1

Re: [Users] Openser as a Outbound Proxy
by Klaus Darilion 04 Jan '06

04 Jan '06

AFAIK you can't do this with (open)ser. regards klaus Gene Willingham wrote: > I would like to see if I can use Openser as an Outbound Proxy for Asterisk. > I would like SIP clients to register with asterisk, but go through Openser > first. > > What I need Openser to do is receive the request, deal with NAT issues then > forward the request to Asterisk for registration. I see two possibilities. > > 1. Openser configured as Outbound proxy setting on the SIP device. > > 2. Openser configured as proxy, but then forward request to asterisk. > > Any help or pointers greatly appreciated. > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Users] Serweb admin login issue
by Antoine Fressancourt 04 Jan '06

04 Jan '06

Hello there, I know the question may have been asked a lot of times, but I am really scratching my head with hat issue : I installed serweb to use with my openser SIP server. I followed the install procedure for serweb, and now I am facing a problem : I can't log on th eserweb admin page with admin / heslo. I have tried all the tricks I could see on the different ailing list discussions. I have tried to remove the domain's autocompletion and tried different possibilities by myself. Could you help me figuring out wher serweb stores the admin login information so that I can change it by hand and try to log in after that ? Thank you for your help Antoine

2 3

[Users] Problem with login on serweb interface
by Antoine Fressancourt 04 Jan '06

04 Jan '06

Hello there, I know the question may have been asked a lot of times, but I am really scratching my head with hat issue : I installed serweb to use with my openser SIP server. I followed the install procedure for serweb, and now I am facing a problem : I can't log on th eserweb admin page with admin / heslo. I have tried all the tricks I could see on the different ailing list discussions. I have tried to remove the domain's autocompletion and tried different possibilities by myself. Could you help me figuring out wher serweb stores the admin login information so that I can change it by hand and try to log in after that ? Thank you for your help Antoine

1 0

[Users] Openser as a Outbound Proxy
by Gene Willingham 04 Jan '06

04 Jan '06

I would like to see if I can use Openser as an Outbound Proxy for Asterisk. I would like SIP clients to register with asterisk, but go through Openser first. What I need Openser to do is receive the request, deal with NAT issues then forward the request to Asterisk for registration. I see two possibilities. 1. Openser configured as Outbound proxy setting on the SIP device. 2. Openser configured as proxy, but then forward request to asterisk. Any help or pointers greatly appreciated.

1 0

Re: Re: [Users] help for openser with tls test
by Kenny Yeh 04 Jan '06

04 Jan '06

Klaus Darilion,您好！ Thans for your help!!!! I am trying..... but my client minisip phone 's setup is follow your tips? what's the ca and what's the private key? as openser's cfg file? ======= 2006-01-04 17:55:20 您在来信中写道：======= >Kenny Yeh wrote: >> users，您好！ >> >> Hi, >> You reply my question.But no answer my question. >> I build a openser with tls support,and client is minisip.I want to test the tls between server from client. >> I do my right cfg file on openser.I donn't know if you want to read it? please see the below: >> cfg file. >> debug=9 # debug level (cmd line: -dddddddddd) >> fork=yes >> log_stderror=no # (cmd line: -E) >> >> check_via=no # (cmd. line: -v) >> dns=no # (cmd. line: -r) >> rev_dns=no # (cmd. line: -R) >> #port=5060 >> children=4 >> fifo="/tmp/openser_fifo" >> #fifo_db_url="mysql:mysql_url" >> >> # >> # uncomment the following lines for TLS support >> disable_tls = 0 >> listen = tls:192.168.2.95:5061 >> tls_verify = 1 >> tls_require_certificate = 0 >> tls_method = SSLv23 >> tls_certificate = "/ca/demoCA/cacert.pem" >> tls_private_key = "/ca/demoCA/private/cakey.pem" >> tls_ca_list = "/ca/openser2/calist.pem" > >Hi Kenny! > >CA is the Caetificae Authority (the organization that signs the >certificates). > >tls_ca_list = "/ca/demoCA/cacert.pem" > >The private key is YOUR private key, not the key of the CA! >tls_private_key = "/ca/openser2/privkey.pem" >The certificate is YOUR certificate, not the certificate of the CA! >tls_certificate = "/ca/openser2/cert.pem" > >further, I'm not sure if minisip supports SSL. Try: >tls_method = TLSv1 > >regards >Klaus >> down is by default. >> >> I can start ser well,and I see the ser.log ,TLS is running,some ca files is loaded.now I start minisip phone to registar,but minisip phone's error messenge:"exception caught where creating tls server" I donn't konw why? >> >> I donn't know if my detail is enough,If you are ok,please send me the steps help to test TLS cfg file or manual book for cookie. >> >> I found the tls help file, >> >> mkdir demoCA >> >> This is the default CA name and it must be exactly as set in your openssl configuration /etc/ss/openssl.cnf : >> >> how to set in openssl.cnf file? or if I didn't ,tls cann't be support? >> >> >> 　　　　　　　　致 >> 礼！ >> >> >> 　　　　　　　　Kenny Yeh >> 上海金叶通讯科技有限公司 >> TEL:8621-6421-6758 ext.311 >> 　　　　　　　　kenny(a)artdio.com.tw >> 　　　　　　　　　　2006-01-04 >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-04

1 0

[Users] help for openser with tls test
by Kenny Yeh 04 Jan '06

04 Jan '06

users，您好！ Hi, You reply my question.But no answer my question. I build a openser with tls support,and client is minisip.I want to test the tls between server from client. I do my right cfg file on openser.I donn't know if you want to read it? please see the below: cfg file. debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 children=4 fifo="/tmp/openser_fifo" #fifo_db_url="mysql:mysql_url" # # uncomment the following lines for TLS support disable_tls = 0 listen = tls:192.168.2.95:5061 tls_verify = 1 tls_require_certificate = 0 tls_method = SSLv23 tls_certificate = "/ca/demoCA/cacert.pem" tls_private_key = "/ca/demoCA/private/cakey.pem" tls_ca_list = "/ca/openser2/calist.pem" down is by default. I can start ser well,and I see the ser.log ,TLS is running,some ca files is loaded.now I start minisip phone to registar,but minisip phone's error messenge:"exception caught where creating tls server" I donn't konw why? I donn't know if my detail is enough,If you are ok,please send me the steps help to test TLS cfg file or manual book for cookie. I found the tls help file, mkdir demoCA This is the default CA name and it must be exactly as set in your openssl configuration /etc/ss/openssl.cnf : how to set in openssl.cnf file? or if I didn't ,tls cann't be support? 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 上海金叶通讯科技有限公司 TEL:8621-6421-6758 ext.311 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-04

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[Users] help for openser with tls
by Kenny Yeh 04 Jan '06

04 Jan '06

users，您好！ I try to build a server with tls support by openser.I want to know how to know openser is with tls support. I modify the config file followed TLS help file.and openser is started well,but how I know it's work right? and my client is minisip,but always send error messenges to me "exception caught when creating tls server",why? I did all the things ,but .... Who can help me,or give me a example. 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 上海金叶通讯科技有限公司 TEL:8621-6421-6758 ext.311 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-04

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[Users] help for TLS on openser
by Kenny Yeh 04 Jan '06

04 Jan '06

users，hello Happly new year! My problem: 1.I start openser with TLS support,follow the TLS help file,create a ca/demoCA/ and create cert.pem,private.pem,key.pem.openser is start well,but I cann't list the listen ports by openserctl ps,how I can be sure openser is started by TLS support? 2.My client is minisip phone that TLS support.but when I start minisip to registar,the error messenge is "exception caught when creating TLS server",I donn't know what's happend. Please help me ! 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 上海金叶通讯科技有限公司 TEL:8621-6421-6758 ext.311 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-04

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[Users] How to write auth username to an AVP (with avpops) for future checking?
by Marcello Lupo 03 Jan '06

03 Jan '06

Hi, probably a stupid question, but how to write the auth username passed on authentication to an AVP?? I tried to access $ua pseudo variable in openser to get it in to the AVP but it failed, obviously.As the docs says the pseudo variables are accessible only for avp_print function. Any help? Thanks, Bye, Marcello

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[Users] Problem with SQL module and usrloc
by Antoine Fressancourt 03 Jan '06

03 Jan '06

Hello, First of all, Happy new year to all the readers of this mailing list ! I have a slightly strange issue with my openser installation. I am using a mySQL database to store the users of my SIP server. I am trying to start the server while the database is on a distant server. The logs are returning this error, and I can't undrestand what went wrong : 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) usrloc:preload_udomain: Wrong version v136410800 for table <aliases>, expected v1001 0(0) register_udomain(): Error while preloading domain 'aliases' 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration Do you have any clue ? Antoine

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[Users] credit card fraud [OT]
by Iqbal 03 Jan '06

03 Jan '06

Hi I know this is off topic, but am sure lots of VoIP companies face this, whats the best strategy to counter this, aside from Visa secure and similar systems. On alarger scale have most just written off the charges and costs incurred, or are there semi-elaborate ways of reducing the risk. I address verification all seems a non-starter unless you wish to block out entire ranges. Any thoughts pointers would be great Iqbal

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Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
by 'Jan Janak' 02 Jan '06

02 Jan '06

Hello, Attached is a patch that implements "swap_direction" parameter of acc module. If you turn the parameter on in the configuration file: modparam("acc", "swap_direction", 1) then the acc module will swap Calling-Station-ID and Called-Station-ID values when necessary (in this case BYE comming from the callee). Put the patch in top level source directory (ser-0.9.x) and type: patch -p0 < swap.patch The patch should work with any 0.9.x release. Note that Alan DeKok is wrong in thinking that this is a bug in SER. This particular problem is a result of incomplete specification of RADIUS use with SIP. I will commit the patch in CVS so it will be included in future SER releases. Jan. On 23-12-2005 17:10, Lenir wrote: > Please read the reply below from one of the maintainers of freeradius: > > "Lenir" <lenirsantiago(a)yahoo.com> wrote: > > But if UserB hangs up on UserA: SER generates a stop-record where the > > Calling-Station-Id is UserB and the Called-Station-Id is UserA, this > > is the undesired and incorrect behavior. > > It would appear to be a bug in SER. > > > To me the Calling-Station-Id and the Called-Station-Id should be the > > same for both start and stop records, am I right by thinking that? > > Yes. > > > According to the developers of SER/OpenSER, this is the correct > > behavior, whoever sends the hangup signal (BYE or CANCEL) is > > considered the Calling-Station-Id, and they are unwilling to modify or > > create a patch to "fix" this. > > What they do for something inside of SER is their business. When they > generate RADIUS packets, they should follow RADIUS standards and > interoperability. The expectation, as you said, is that the > Calling/Called-Station-Id doesn't change during a session. If it does, it's > a bug and they should fix it. > > Alan DeKok. > - > List info/subscribe/unsubscribe? See > http://www.freeradius.org/list/users.html > > > > -----Original Message----- > From: 'Jan Janak' [mailto:jan@iptel.org] > Sent: Thursday, November 17, 2005 10:33 AM > To: Lenir > Cc: 'Klaus Darilion'; serdev(a)iptel.org; serusers(a)iptel.org; > devel(a)openser.org; users(a)openser.org > Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius > Accounting > > On 17-11-2005 10:21, Lenir wrote: > > In this case the radius proxy wont work, because you never can anticipate > > who hangs up the call, thus radius wont know who hung up the > call...Besides > > all other voice applications/hardware (SIP and H323) that use radius do > not > > behave like that, the Called-Station-ID ALWAYS remains the same, as with > the > > Calling-Station-ID. > > Could you name those SIP applications that behave the way you describe ? > > Jan. >

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[Users] Dealing with reinvites
by Matt Schulte 02 Jan '06

02 Jan '06

All, We have had a new issue arise within openser, basically we connect to a SIP carrier that uses Asterisk as a proxy. Naturally, this carrier is hellbent on offloading the RTP traffic, thus the reinvites. My issue is that they appear to be sending the second invite "too fast", before the initial RTP even gets a chance to establish. This was working just fine before, and I believe they changed something. Anyway, they're refusing to help with this issue, so now I must "fix" it on my end. OpenSER is in-part, the problem.. See below: (Sorry if formatting is off, dump from tethereal ..) 0.000000 204.13.233.13 -> 206.80.70.47 SIP/SDP Request: INVITE sip:8886963856@206.80.70.47, with session description 0.001752 206.80.70.47 -> 204.13.233.13 SIP Status: 100 trying -- your call is important to us 0.002723 206.80.70.47 -> 206.80.70.54 SIP/SDP Request: INVITE sip:+1314xxxxxxx@sip.stl.netlogic.net, with session description 0.010327 206.80.70.54 -> 206.80.70.47 SIP Status: 100 Trying 0.292175 206.80.70.54 -> 206.80.70.47 SIP Status: 180 Ringing 0.292692 206.80.70.47 -> 204.13.233.13 SIP Status: 180 Ringing 2.016246 206.80.70.54 -> 206.80.70.47 SIP/SDP Status: 200 OK, with session description 2.016893 206.80.70.47 -> 204.13.233.13 SIP/SDP Status: 200 OK, with session description 2.050871 204.13.233.13 -> 206.80.70.47 SIP Request: ACK sip:+1314xxxxxxx@206.80.70.54 2.051233 204.13.233.13 -> 206.80.70.47 SIP/SDP Request: INVITE sip:+1314xxxxxxx@206.80.70.54 2.051827 206.80.70.47 -> 204.13.233.13 SIP Status: 100 trying --<<KABOOM>>-- 2.051956 206.80.70.47 -> 206.80.70.54 SIP/SDP Request: INVITE sip:+1314xxxxxxx@206.80.70.54, 2.052112 206.80.70.47 -> 206.80.70.54 SIP Request: ACK sip:+13142664000@206.80.70.54 2.053028 206.80.70.54 -> 206.80.70.47 SIP/SDP Status: 200 OK, with session description 2.053336 206.80.70.54 -> 206.80.70.47 SIP Status: 503 Server error Where you see "kaboom", is the problem. If you notice, the time between the first ACK and second INVITE is about 1ms .. While this shouldn't be a problem, you'll notice when the ACK/INVITES are getting statefully forwarded, they're getting sent out of order. The end point is an Asterisk machine, thus the 503 error. Carrier_Ast --> OpenSER --> Netlogic_Ast I am using loose_route to forward, just like everyone else, they are even hitting loose_route in the correct order.. Thoughts? Suggestions? This seems like something internal to OpenSER, I have tried butchering the config to force the ACK out first and it just created more problems. Thanks! Matt S

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[Users] trusted ips and domain in radius
by Jayesh Nambiar 02 Jan '06

02 Jan '06

hi all, I am doing all my accounting and authorization in radius. I wanted to ask if there is a way of authenticating src_ips through radius like using the allow_trusted function from mysql. pls suggest me a way of authenticating src_ips via radius. I also wanted multidomain support, but the domain module also depends on a database module. My idea was to do everything using radius. Can someone help me in this regard. Any suggestions would help me a lot. thanx in advance. jayesh Send instant messages to your online friends http://in.messenger.yahoo.com

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[Users] handling of command line parameters
by Klaus Darilion 31 Dec '05

31 Dec '05

Hi! Is this observation correct? 1. If an option is defined in the openser.cfg file, this definition will be used. 2. In an option is not defined in the openser.cfg file, the default value will be used, except the option is defined via a command line parameter. regards klaus

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[Users] invite timers
by Mike Williams 31 Dec '05

31 Dec '05

I've run into an interesting issue with invite timers: when a user makes an outbound call, to say, a PSTN gateway, it would be best if the OpenSER fr_invite_timer was set very high. However, this timer controls how long I want the call to ring for before timing out and sending users to my local voicemail server.I would like the timer to be set lower for this. Has anyone come up with a creative way around this? Am I missing something obvious? All help is appreciated. ---Mike

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[Users] help: anonymous call
by unplug 30 Dec '05

30 Dec '05

Below is the anonymous call invite method compared with the normal call. In the from header, there is missing username. How can I add an anonymous username (i.e. <sip:anonymous@203.193.46.4>) in the from header if the call is from anonymous? Invite method from Anonymous call ---------------------------------------------- U 203.193.46.4:54169 -> 203.193.46.6:5060 INVITE sip:36428473@203.193.46.6:5060 SIP/2.0. Via: SIP/2.0/UDP 203.193.46.4:5060. From: <sip:203.193.46.4>;tag=1924AB7C-EE5. <----------------------------- missing username To: <sip:36428473@203.193.46.6>. Date: Wed, 28 Dec 2005 09:22:19 GMT. Invite method from normal call ---------------------------------------------- U 203.193.46.4:50197 -> 203.193.46.6:5060 INVITE sip:36428473@203.193.46.6:5060 SIP/2.0. Via: SIP/2.0/UDP 203.193.46.4:5060. From: <sip:34263634@203.193.46.4>;tag=978CF54-291. To: <sip:36428473@203.193.46.6>. Date: Fri, 23 Dec 2005 06:29:08 GMT.

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[Users] Media Proxy
by Fernando Rodriguez 29 Dec '05

29 Dec '05

Is there any manual on how to setup multiple mediaproxy instances in one machine ?? For different network cards and different lans?? Thanks Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

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[Users] Disabling rtpproxy or mediaproxy when both SIP UA are behind same NAT
by Rafael Risco GV 29 Dec '05

29 Dec '05

Hi I think I am very close to solve this issue, based on suggestions of Tavis and Klaus for openser in: http://openser.org/dokuwiki/doku.php?id=how_to_set_up_nathelper_rtpproxy_wh… I´ve been able to disable rtpproxy in this cases but I have a problem with accounting, if the callee hungup first appears an error and no cdr for BYE is generated until caller hungup the call, bellow the errors and config. thanks for any help rafael 4(22193) BYE - STOP ACCOUNTING 4(22193) DEBUG: add_param: tag=3658302912 4(22193) grep_sock_info - checking if host==us: 13==13 && [192.168.1.205] == [10.0.2.130] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) grep_sock_info - checking if host==us: 13==9 && [192.168.1.205] == [127.0.0.1] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) check_self: host != me 4(22193) grep_sock_info - checking if host==us: 13==13 && [10.0.2.130] == [10.0.2.130] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) check_via_address(10.0.1.25, 192.168.1.100, 0) 4(22193) ERROR: udp_send: sendto(sock,0x4051e800,588,0,0x4051f2ac,16): Operation not permitted(1) 4(22193) msg_send: ERROR: udp_send failed 4(22193) ERROR:tm:t_forward_nonack: sending request failed 4(22193) DEBUG: add_to_tail_of_timer[4]: 0x4051f2cc 4(22193) DEBUG: add_to_tail_of_timer[0]: 0x4051f2dc 4(22193) ERROR:tm:t_relay_to: t_forward_nonack returned error 4(22193) parse_headers: flags=ffffffffffffffff 4(22193) check_via_address(10.0.1.25, 192.168.1.100, 0) 4(22193) WARNING:vqm_resize: resize(0) called . . .

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[Users] Disabling rtpproxy or mediaproxy when both SIP UA are behind same NAT
by Rafael Risco GV 29 Dec '05

29 Dec '05

Hi I think I am very close to solve this issue, based on suggestions of Tavis and Klaus for openser in: http://openser.org/dokuwiki/doku.php?id=how_to_set_up_nathelper_rtpproxy_wh… I´ve been able to disable rtpproxy in this cases but I have a problem with accounting, if the callee hungup first appears an error and no cdr for BYE is generated until caller hungup the call, bellow the errors and config. thanks for any help rafael 4(22193) BYE - STOP ACCOUNTING 4(22193) DEBUG: add_param: tag=3658302912 4(22193) grep_sock_info - checking if host==us: 13==13 && [192.168.1.205] == [10.0.2.130] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) grep_sock_info - checking if host==us: 13==9 && [192.168.1.205] == [127.0.0.1] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) check_self: host != me 4(22193) grep_sock_info - checking if host==us: 13==13 && [10.0.2.130] == [10.0.2.130] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) check_via_address(10.0.1.25, 192.168.1.100, 0) 4(22193) ERROR: udp_send: sendto(sock,0x4051e800,588,0,0x4051f2ac,16): Operation not permitted(1) 4(22193) msg_send: ERROR: udp_send failed 4(22193) ERROR:tm:t_forward_nonack: sending request failed 4(22193) DEBUG: add_to_tail_of_timer[4]: 0x4051f2cc 4(22193) DEBUG: add_to_tail_of_timer[0]: 0x4051f2dc 4(22193) ERROR:tm:t_relay_to: t_forward_nonack returned error 4(22193) parse_headers: flags=ffffffffffffffff 4(22193) check_via_address(10.0.1.25, 192.168.1.100, 0) 4(22193) WARNING:vqm_resize: resize(0) called . . . # ----------- global configuration parameters ------------------------ #/* Uncomment these lines to enter debugging mode debug=6 fork=yes log_stderror=yes #*/ listen=10.0.2.130 listen=127.0.0.1 port=5060 # hostname matching an alias will satisfy the condition uri==myself". alias=mydomain.com.pe alias=10.0.2.130 alias=127.0.0.1 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 fifo="/tmp/openser_fifo" fifo_mode=0666 # Fifo permissions can be changes from here. # sip_warning - Should replies include extensive warnings? # By default yes, it is good for trouble-shooting. sip_warning=yes # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" # digest authentication loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/local/lib/openser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "mysql://admin:heslo@localhost/openser") modparam("usrloc", "use_domain", 1) modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # ------------- Mysql Accounting parameters modparam("acc", "log_flag", 1) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://admin:heslo@localhost/openser") modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsputdr") modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 40 ) # Timer which hits if no final reply for an INVITE modparam("tm", "wt_timer", 20 ) # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("group", "db_url", "mysql://seradmin:heslo@localhost/openser") modparam("uri_db", "db_url", "mysql://seradmin:heslo@localhost/openser") # ------------- registration parameters modparam("registrar", "nat_flag", 6) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 86400) modparam("registrar", "default_expires", 3600) modparam("registrar", "desc_time_order", 1) modparam("registrar", "append_branches", 1) modparam("registrar", "use_domain", 1) # !! Nathelper # modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # -------------------------- request routing logic -------------------------- route { log(1, "-------------------------------------------\n"); log(1, "entering main loop\n"); # 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"); return; }; if ( msg:len >= max_len ) { sl_send_reply("513", "Message too big"); return; }; # set flag for Radius Accounting: if (!method=="OPTIONS") setflag(3); if (method=="INVITE") { log(1, "INVITE MESSAGE RECEIVED - START ACC\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE") { log (1, "BYE - STOP ACCOUNTING\n"); setflag(1); }; if (method=="CANCEL") { log (1, "CANCEL - STOP ACCOUNTING\n"); setflag(1); }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- # 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(); if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route() && (!src_ip==10.0.2.130 && !src_port==5070)) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(7); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(2); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!uri==myself && (!src_ip==10.0.2.130 && !src_port==5070)) { route(1); return; }; if (method=="CANCEL") { route(2); return; }; if (method == "REGISTER") { if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); log(1, "ANALYZING REGISTER REQUEST\n"); # ... to use digest authentication if (is_user_in("Request-URI", "desactivado")) { sl_send_reply("402", "Su cuenta fue desactivada por falta de pago"); return; }; if (!www_authorize("", "subscriber")) { log(1," ----- Fails to Register \n"); www_challenge("", "0"); return; }; # only signed users are allowed if (!check_to()) { log(1, "LOG: Hijack!!!--> unsigned user registration attempt\n"); sl_send_reply("403", "hijack attempt!!!! Only signed users are allowed"); return; }; consume_credentials(); log(1," Registered!!! \n"); if (!save("location")) { sl_reply_error(); }; return; }; # INVITE ? First check the source of the call #******************************************** # If the call comes from the gateways, no authentication is required. if (method == "INVITE" && (src_ip==10.0.2.145 || src_ip==10.0.2.132|| src_ip== 10.0.2.131)) { log(1,"Call from pstn|*, no authentication is required. \n"); # If the call comes from B2BUA, no authentication is required. # The first leg of the call has already been authenticated. } else if (src_ip==10.0.2.130 && src_port==5070) { log(1,"Call from B2BUA, no authentication is required. \n"); } else { log(1, "ANALYZING INVITE REQUESTs\n"); if (method == "INVITE" && !src_ip==10.0.2.130 && !src_port==5070){ if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); return; } else { if (!check_from()) { sl_send_reply("403", "Only registered users are allowed"); log(1," ----> Only registered users are allowed \n"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(7); }; }; # Not all the users are PREPAID, so we check the database # to see if the call will be routed through B2BUA. # If every call is to be routed through B2BUA the "is_user_in" # conditional is not required. # Do not use b2bua for local calls (660++++) if (is_user_in("From", "prepaidb") && uri=~"^sip:00") { log(1," ----> Usuario PREPAGO!!! enviando a b2bua... \n"); rewritehostport("10.0.2.130:5070"); t_relay_to_udp("10.0.2.130", "5070"); return; }; }; }; # End of if (method == "INVITE" |... /* *********** Dial out to Local and PSTN logic ********* */ if(uri=~"^sip:00"){ log(1,"00N match - Larga Distancia Internacional \n"); if (!is_user_in("from", "ldix")) { log(1,"No permission for international calls \n"); sl_send_reply("403", "No permission for international calls"); acc_db_request("403 Forbidden", "missed_calls"); return; }; rewritehostport("10.0.2.131:5070"); strip(2); route(1); return; }; /* ******************************************************************** */ lookup("aliases"); if (uri!=myself) { route(1); return; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (is_user_in("Request-URI", "voicemail")) { log(1, "requested user is in voicemail group \n"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { log(1,"unable to locate user X ... sending to route(4)! \n"); # handle user which was not found route(4); return; }; ### Check if UAS are behind the same NAT system: ### if (isflagset(6) && isflagset(7)){ log(2, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("i:450", "$dd"); if (avp_check("i:450", "eq/$src_ip/g")){ log(3, "Detected Two Clients Behind the Same NAT - Disabling Mediaproxy"); # Do not use media-proxy as the clients seem to be behind the same NAT resetflag(6); resetflag(7); route(2); return; }; }; route(1); # if user is on-line and is in Voicemail group, enable redirection if (method == "INVITE" && isflagset(4)) { log(1, "invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); }; } ### ##### ####### ########## ########### ############### ################# ##################### route[1] { # if client or server know to be behind a NAT, enable relay if (isflagset(6) || isflagset(7)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" && (isflagset(6) || isflagset(7))) { unforce_rtp_proxy(); }; sl_reply_error(); }; log(1, "Route[1]: Send it out now!!!\n"); } # !! Nathelper onreply_route[1] { # NATed transaction ? # Not all 2xx messages have a content body so here we # make sure our Content-Length > 0 to avoid a parse error if ((isflagset(6) || isflagset(7)) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); if (!search("^Content-Length:\ 0")) { force_rtp_proxy(); log(1, "onreply_route1: force rtp proxy!!!\n"); }; # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } # -------------- Default Message Handler --------------------- route[2]{ # ---------------------------------------------------- # Default Message Handler # ---------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(7)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } # --------------- Handling of Unavailable user ---------------- route[4] { # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" || method=="BYE" || method=="OPTIONS")) { sl_send_reply("404", "Not Found"); acc_db_request("404 Not Found", "missed_calls"); log(1, "acc 404 Not Found 1 \n"); return; }; # not voicemail subscriber if (!isflagset(4) && !method=="OPTIONS" && !method=="ACK" && !method=="BYE") { sl_send_reply("404", "Not Found and no voicemail turned on !! "); acc_db_request("404 Not Found", "missed_calls"); log(1, " acc 404 Not Found and no voicemail \n"); return; }; # forward to voicemail adding prefix to simplify asterisk " extension.conf" :: # exten => _vmXXXXXXX,1,SetLanguage(es) # exten => _vmXXXXXXX,2,Voicemail(u${EXTEN:2}) # exten => _vmXXXXXXX,3,Hangup prefix("vm"); acc_db_request("404 Not Found -> Vm", "missed_calls"); rewritehostport("10.0.2.131:5070"); t_relay_to_udp("10.0.2.131", "5070"); } # If forwarding downstream did not succeed, try voicemail running at Asterisk failure_route[1]{ if (t_check_status("408")){ # revert_uri (); back to the original URI, makes me loose all lookup/rewrite stuff prefix("vm"); rewritehostport ("10.0.2.131:5070"); acc_db_request("408 Timeout -> Vm", "missed_calls"); append_branch(); t_relay(); return; } else if (t_check_status("486")){ # revert_uri (); back to the original URI, makes me loose all lookup/rewrite stuff prefix("vm"); rewritehostport ("10.0.2.131:5070"); acc_db_request("486 Busy -> Vm", "missed_calls"); append_branch(); t_relay(); return; } }

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[Users] Mediaproxy
by Paul van Schagen 29 Dec '05

29 Dec '05

Gents, Can anybody tell me if mediaproxy works for far-end Nat traversal for symmetric firewalls ? Thanks ahead. Paul vna Schagen

2 1

[Users] Src&Dest ip avpops
by Fernando Rodriguez 29 Dec '05

29 Dec '05

Hello, Is there a place were i can read about avpops module regarding src and dst ip this is for using mediaproxy when the users are in diferent interfaces of the equipment or diferent networks so i can only mediaproxy the ones that need this functionalyty and the ones on the same network connect direct from one to the other. Thanks a lot Fernando Rodriguez V.

3 3

Re: [Users] RE: [Serusers] Distributing "OSP enable SER" binary
by Dmitry Isakbayev 28 Dec '05

28 Dec '05

Hi Elton, Some documentation is available @ http://www.sipfoundry.org/OSP. You can try forwarding your question to osp(a)list.sipfoundry.org (https://list.sipfoundry.org/mailman/listinfo/osp) Regards, Dmitry > From: "Elton Machado" <elton.machado(a)gmail.com> > Subject: [Users] RE: [Serusers] Distributing "OSP enable SER" binary > file w/ source code > To: <vmathur(a)transnexus.com> > Cc: serusers(a)iptel.org, "'users openser.org'" <users(a)openser.org> > Message-ID: <43b1c85f.1e92b00c.4a45.ffffca62(a)mx.gmail.com> > Content-Type: text/plain; charset="us-ascii" > > Hi, > > After some hours around OpenOSP I did it to compile under FreeBSD ;) > > So.. Next step was to try to find documentation about OpenOSP Server. > Problem iscould find much not to say anything. > Everything I found is about the client and about openser and asterisk > modules. > > Can you provide me some sort of configuration for the server? > > I'm trying to using it with Openser 1.0.0 and my system is running Freebsd > 6.0 > > I had a hard time to get it under it as it request some changes in code > and > also very old stuff, it is possible to have it cooworking with openldad > 2.3 > and a newer openssl an berkley db? > > So any information would be usefull as I can't figure out much with > documentation avaible I'm more insterested in OpenOSP Server side as I > have > clear information about client side. > > Regards, > > Elton Machado >

1 0

[Users] Disabling rtpproxy or mediaproxy when both SIP UA are behind same NAT
by Rafael Risco GV 28 Dec '05

28 Dec '05

Hi I think I am very close to solve this issue, based on suggestions of Tavis and Klaus for openser in: http://openser.org/dokuwiki/doku.php?id=how_to_set_up_nathelper_rtpproxy_wh… I´ve been able to disable rtpproxy in this case but I have a problem with accounting, if the callee hungup first appears an error and no cdr for BYE is generated until caller hungup the call, bellow the errors and config. thanks for any help rafael error when callee hungup tha call: 2(21685) entering main loop 2(21685) BYE - STOP ACCOUNTING 2(21685) ERROR: udp_send: sendto(sock,0x4051ea70,588,0,0x4051f27c,16): Operation not permitted(1) 2(21685) msg_send: ERROR: udp_send failed 2(21685) ERROR:tm:t_forward_nonack: sending request failed more detailed (debug=6) ::: 4(22193) ------------------------------------------- 4(22193) entering main loop 4(22193) parse_headers: flags=100 4(22193) DEBUG: add_param: tag=1184914029 4(22193) DEBUG:parse_to:end of header reached, state=29 4(22193) DEBUG: get_hdr_field: <To> [56]; uri=[sip:6606060@mydomain.com.pe] 4(22193) DEBUG: to body [Rafo <sip:6606060@mydomain.com.pe>] 4(22193) get_hdr_field: cseq <CSeq>: <54121> <BYE> 4(22193) DEBUG:maxfwd:is_maxfwd_present: value = 70 4(22193) BYE - STOP ACCOUNTING 4(22193) DEBUG: add_param: tag=3658302912 4(22193) DEBUG:parse_to:end of header reached, state=29 4(22193) parse_headers: flags=40 4(22193) parse_headers: flags=200 4(22193) is_preloaded: No 4(22193) grep_sock_info - checking if host==us: 13==13 && [192.168.1.205] == [10.0.2.130] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) grep_sock_info - checking if host==us: 13==9 && [192.168.1.205] == [127.0.0.1] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) grep_sock_info - checking if host==us: 13==13 && [192.168.1.205] == [10.0.2.130] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) grep_sock_info - checking if host==us: 13==9 && [192.168.1.205] == [127.0.0.1] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) check_self: host != me 4(22193) grep_sock_info - checking if host==us: 13==13 && [10.0.2.130] == [10.0.2.130] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) after_loose: Topmost route URI: 'sip:10.0.2.130;ftag=1184914029;lr=on' is me 4(22193) parse_headers: flags=200 4(22193) DEBUG: get_hdr_body : content_length=0 4(22193) found end of header 4(22193) find_next_route: No next Route HF found 4(22193) after_loose: No next URI found 4(22193) DEBUG: has_totag: totag found 4(22193) DEBUG: t_newtran: msg id=12 , global msg id=5 , T on entrance=0xffffffff 4(22193) parse_headers: flags=ffffffffffffffff 4(22193) parse_headers: flags=78 4(22193) t_lookup_request: start searching: hash=46346, isACK=0 4(22193) DEBUG: RFC3261 transaction matching failed 4(22193) DEBUG: t_lookup_request: no transaction found 4(22193) DBG: trans=0x4051f1b0, callback type 1, id 0 entered 4(22193) parse_headers: flags=58 4(22193) DEBUG: mk_proxy: doing DNS lookup... 4(22193) check_via_address(10.0.1.25, 192.168.1.100, 0) 4(22193) ERROR: udp_send: sendto(sock,0x4051e800,588,0,0x4051f2ac,16): Operation not permitted(1) 4(22193) msg_send: ERROR: udp_send failed 4(22193) ERROR:tm:t_forward_nonack: sending request failed 4(22193) DEBUG: add_to_tail_of_timer[4]: 0x4051f2cc 4(22193) DEBUG: add_to_tail_of_timer[0]: 0x4051f2dc 4(22193) ERROR:tm:t_relay_to: t_forward_nonack returned error 4(22193) parse_headers: flags=ffffffffffffffff 4(22193) check_via_address(10.0.1.25, 192.168.1.100, 0) 4(22193) WARNING:vqm_resize: resize(0) called 4(22193) DBG: trans=0x4051f1b0, callback type 128, id 0 entered 4(22193) DEBUG: cleanup_uac_timers: RETR/FR timers reset 4(22193) DEBUG: add_to_tail_of_timer[2]: 0x4051f1f8 4(22193) DEBUG:tm:_reply_light: reply sent out. buf=0x8136738: SIP/2.0 4..., shmem=0x40520728: SIP/2.0 4 4(22193) DEBUG:tm:_reply_light: finished 4(22193) ERROR: generation of a stateful reply on error succeeded 4(22193) DEBUG:destroy_avp_list: destroying list (nil) 4(22193) receive_msg: cleaning up 2(22191) udp_rcv_loop: probing packet received from 10.0.1.25 54330 10(22199) DEBUG: timer routine:0,tl=0x4051f2dc next=(nil) 10(22199) DEBUG: timer routine:4,tl=0x4051f2cc next=(nil) 4(22193) SIP Request: 4(22193) method: <REGISTER> 4(22193) uri: <sip:mydomain.com.pe> 4(22193) version: <SIP/2.0> 4(22193) parse_headers: flags=2 4(22193) Found param type 232, <branch> = <z9hG4bK08ac1f39>; state=6 4(22193) Found param type 235, <rport> = <n/a>; state=17 4(22193) end of header reached, state=5 4(22193) parse_headers: Via found, flags=2 4(22193) parse_headers: this is the first via 4(22193) After parse_msg... 4(22193) preparing to run routing scripts... 4(22193) ------------------------------------------- # ----------- global configuration parameters ------------------------ #/* Uncomment these lines to enter debugging mode debug=6 fork=yes log_stderror=yes #*/ listen=10.0.2.130 listen=127.0.0.1 port=5060 # hostname matching an alias will satisfy the condition uri==myself". alias=mydomain.com.pe alias=10.0.2.130 alias=127.0.0.1 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 fifo="/tmp/openser_fifo" fifo_mode=0666 # Fifo permissions can be changes from here. # sip_warning - Should replies include extensive warnings? # By default yes, it is good for trouble-shooting. sip_warning=yes # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" # digest authentication loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/local/lib/openser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "mysql://admin:heslo@localhost/openser") modparam("usrloc", "use_domain", 1) modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # ------------- Mysql Accounting parameters modparam("acc", "log_flag", 1) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://admin:heslo@localhost/openser") modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsputdr") modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 40 ) # Timer which hits if no final reply for an INVITE modparam("tm", "wt_timer", 20 ) # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("group", "db_url", "mysql://seradmin:heslo@localhost/openser") modparam("uri_db", "db_url", "mysql://seradmin:heslo@localhost/openser") # ------------- registration parameters modparam("registrar", "nat_flag", 6) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 86400) modparam("registrar", "default_expires", 3600) modparam("registrar", "desc_time_order", 1) modparam("registrar", "append_branches", 1) modparam("registrar", "use_domain", 1) # !! Nathelper # modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # -------------------------- request routing logic -------------------------- route { log(1, "-------------------------------------------\n"); log(1, "entering main loop\n"); # 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"); return; }; if ( msg:len >= max_len ) { sl_send_reply("513", "Message too big"); return; }; # set flag for Radius Accounting: if (!method=="OPTIONS") setflag(3); if (method=="INVITE") { log(1, "INVITE MESSAGE RECEIVED - START ACC\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE") { log (1, "BYE - STOP ACCOUNTING\n"); setflag(1); }; if (method=="CANCEL") { log (1, "CANCEL - STOP ACCOUNTING\n"); setflag(1); }; if (method=="ACK") log(1, "ACK message received\n"); if (method=="SUBSCRIBE") log(1, "SUBSCRIBE message received\n"); if (method=="NOTIFY") log(1, "NOTIFY message received\n"); if (method=="OPTIONS") log(1, "OPTIONS MESSAGE RECEIVED\n"); if (method=="INFO") log(1, "INFO message received\n"); if (method=="MESSAGE") log(1, "MESSAGE message received\n"); if (method=="REFER") log(1, "REFER message received\n"); # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- # 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(); if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route() && (!src_ip==10.0.2.130 && !src_port==5070)) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(7); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(2); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!uri==myself && (!src_ip==10.0.2.130 && !src_port==5070)) { route(1); return; }; if (method=="CANCEL") { route(2); return; }; if (method == "REGISTER") { if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); log(1, "ANALYZING REGISTER REQUEST\n"); # ... to use digest authentication if (is_user_in("Request-URI", "desactivado")) { sl_send_reply("402", "Su cuenta fue desactivada por falta de pago"); return; }; if (!www_authorize("", "subscriber")) { log(1," ----- Fails to Register \n"); www_challenge("", "0"); return; }; # only signed users are allowed if (!check_to()) { log(1, "LOG: Hijack!!!--> unsigned user registration attempt\n"); sl_send_reply("403", "hijack attempt!!!! Only signed users are allowed"); return; }; consume_credentials(); log(1," Registered!!! \n"); if (!save("location")) { sl_reply_error(); }; return; }; # INVITE ? First check the source of the call #******************************************** # If the call comes from the gateways, no authentication is required. if (method == "INVITE" && (src_ip==10.0.2.145 || src_ip==10.0.2.132|| src_ip== 10.0.2.131)) { log(1,"Call from pstn|*, no authentication is required. \n"); # If the call comes from B2BUA, no authentication is required. # The first leg of the call has already been authenticated. } else if (src_ip==10.0.2.130 && src_port==5070) { log(1,"Call from B2BUA, no authentication is required. \n"); } else { log(1, "ANALYZING INVITE REQUESTs\n"); if (method == "INVITE" && !src_ip==10.0.2.130 && !src_port==5070){ if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); return; } else { if (!check_from()) { sl_send_reply("403", "Only registered users are allowed"); log(1," ----> Only registered users are allowed \n"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(7); }; }; # Not all the users are PREPAID, so we check the database # to see if the call will be routed through B2BUA. # If every call is to be routed through B2BUA the "is_user_in" # conditional is not required. # Do not use b2bua for local calls (660++++) if (is_user_in("From", "prepaidb") && uri=~"^sip:00") { log(1," ----> Usuario PREPAGO!!! enviando a b2bua... \n"); rewritehostport("10.0.2.130:5070"); t_relay_to_udp("10.0.2.130", "5070"); return; }; }; }; # End of if (method == "INVITE" |... /* *********** Dial out to Local and PSTN logic ********* */ if(uri=~"^sip:001"){ log(1,"00N match - Larga Distancia Internacional a USA \n"); if (!is_user_in("from", "ldiusa")) { log(1,"No permission for international calls to USA \n"); sl_send_reply("403", "No permission for international calls to USA"); acc_db_request("403 Forbidden", "missed_calls"); return; }; rewritehostport("10.0.2.131:5070"); strip(2); route(1); return; }; if(uri=~"^sip:00"){ log(1,"00N match - Larga Distancia Internacional \n"); if (!is_user_in("from", "ldix")) { log(1,"No permission for international calls \n"); sl_send_reply("403", "No permission for international calls"); acc_db_request("403 Forbidden", "missed_calls"); return; }; rewritehostport("10.0.2.131:5070"); strip(2); route(1); return; }; /* ******************************************************************** */ lookup("aliases"); if (uri!=myself) { route(1); return; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (is_user_in("Request-URI", "voicemail")) { log(1, "requested user is in voicemail group \n"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { log(1,"unable to locate user X ... sending to route(4)! \n"); # handle user which was not found route(4); return; }; ### if UAS are behind the same NAT system: ### if (isflagset(6) && isflagset(7)){ log(2, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("i:450", "$dd"); if (avp_check("i:450", "eq/$src_ip/g")){ log(3, "Detected Two Clients Behind the Same NAT - Disabling Mediaproxy"); # Do not use media-proxy as the clients seem to be behind the same NAT resetflag(6); resetflag(7); route(2); return; }; }; route(1); # if user is on-line and is in Voicemail group, enable redirection if (method == "INVITE" && isflagset(4)) { log(1, "invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); }; } ### ##### ####### ########## ########### ############### ################# ##################### route[1] { # if client or server know to be behind a NAT, enable relay if (isflagset(6) || isflagset(7)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" && (isflagset(6) || isflagset(7))) { unforce_rtp_proxy(); }; sl_reply_error(); }; log(1, "Route[1]: Send it out now!!!\n"); } # !! Nathelper onreply_route[1] { # NATed transaction ? # Not all 2xx messages have a content body so here we # make sure our Content-Length > 0 to avoid a parse error if ((isflagset(6) || isflagset(7)) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); if (!search("^Content-Length:\ 0")) { force_rtp_proxy(); log(1, "onreply_route1: force rtp proxy!!!\n"); }; # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } # -------------- Default Message Handler --------------------- route[2]{ # ---------------------------------------------------- # Default Message Handler # ---------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(7)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } # --------------- Handling of Unavailable user ---------------- route[4] { # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" || method=="BYE" || method=="OPTIONS")) { sl_send_reply("404", "Not Found"); acc_db_request("404 Not Found", "missed_calls"); log(1, "acc 404 Not Found 1 \n"); return; }; # not voicemail subscriber if (!isflagset(4) && !method=="OPTIONS" && !method=="ACK" && !method=="BYE") { sl_send_reply("404", "Not Found and no voicemail turned on !! "); acc_db_request("404 Not Found", "missed_calls"); log(1, " acc 404 Not Found and no voicemail \n"); return; }; # forward to voicemail adding prefix to simplify asterisk " extension.conf" :: # exten => _vmXXXXXXX,1,SetLanguage(es) # exten => _vmXXXXXXX,2,Voicemail(u${EXTEN:2}) # exten => _vmXXXXXXX,3,Hangup prefix("vm"); acc_db_request("404 Not Found -> Vm", "missed_calls"); rewritehostport("10.0.2.131:5070"); t_relay_to_udp("10.0.2.131", "5070"); } # If forwarding downstream did not succeed, try voicemail running at Asterisk failure_route[1]{ if (t_check_status("408")){ # revert_uri (); back to the original URI, makes me loose all lookup/rewrite stuff prefix("vm"); rewritehostport ("10.0.2.131:5070"); acc_db_request("408 Timeout -> Vm", "missed_calls"); append_branch(); t_relay(); return; } else if (t_check_status("486")){ # revert_uri (); back to the original URI, makes me loose all lookup/rewrite stuff prefix("vm"); rewritehostport ("10.0.2.131:5070"); acc_db_request("486 Busy -> Vm", "missed_calls"); append_branch(); t_relay(); return; } }

1 0

[Users] multiple aliases
by Mike Williams 28 Dec '05

28 Dec '05

Is it possible to use aliases to route to multiple sip uri's?I would like to have one user resolve to say, three different users from the database. Also, I assume I can alias a local sip address to a user in another domain / at another server? I want to have a virtual number that rings three seperate phones when called, regardless of whether they are sip or pstn. Ideally, I would be able to do this with aliases. Thanks for any help. Mike Williams

2 1

[Users] openser with radius problem
by Jayesh Nambiar 28 Dec '05

28 Dec '05

hi all, I tried compiling openser with radius support. everything was fine. I did it according to radius howto mentioned in the iptel.org. The test for authentication was also ok. but when i try to start the openserctl, I get the following error in log: load_module: could not open module </usr/local/lib/openser/modules/acc.so>: libradiusclient-ng.so.2: cannot open shared object file: No such file or directory I googled old mailing list and found out that the LD_LIBRARY_PATH needs to be set. But i have no idea how and where to set this. can somebody pls help me understand this problem. thanx in advance jayesh Send instant messages to your online friends http://in.messenger.yahoo.com

2 1

[Users] caller display
by unplug 28 Dec '05

28 Dec '05

I am using an alias mapped to an account. I want the alias to show in the caller display instead of the real account. Is it possible to configure openser for that purpose, how?

2 2

Re: [Serusers] Distributing "OSP enable SER" binary file w/ source code
by vmathur＠transnexus.com 27 Dec '05

27 Dec '05

Hernan, OSP uses Public key based authentication and encrytion schemes, which are stronger than Radius's shared secrets. If you use OSP for authentication and authorization, you need not use Radius. OSP, just like Radius, has a client stack, which is implemented in the SER, and has a Server, which provides centralized call routing, accounting, and security. At the time of startup, the OSP server and the client, the SER in this case, exchange their public keys. The public and private keys can then be used to encode/decode messages as desired. A typical call setup procedure looks like this 1. The source SER goes to the osp server to get the SIP URI corresopnding to the destination. The message is encoded using the source SER's private key. 2. The osp server decodes the message using source SERs public key and after successful decoding (authentication) returns the route back to the source. Along with the route, it also sends back a digitally signed (using the OSP Servers private key) token. 3. The Source uses the route returned by the OSP Srver to send an INVITE. The INVITE message contains the token issued by the OSP Server 4. The destination decodes/validates the token using the OSP Servers public key. Upon successful validation (authorization) it accepts the call. This saves you the effort of mantaining cumbersome access lists for authentication. I can provide you with more documentatin on OSP and how to use SER with OSP if you wish Thanks, Vikrant ------------------------------------------------------------------------------------------------------------- Fogive my ignorance for I have never heard of OSP before ;) You mentioned Radius in your message. How does Radius authentication work in OSP? I am having a tough time getting mine to work. hernan vmathur(a)transnexus.com wrote: Dear All, I have recently implemented OSP w/ SER. OSP is an ETSI defined protocol, which I am using for ceneralised routing, and security of my inter-domain calls. The problem, however, is that the build process is a little lengthy. I want to contribute my implementation to this group so that anyone who is struggling with SER routing configurations or Radius authentication issues may benefit from it. I was, thus, wondering if we can have a binary file of the OSP enabled SER, that can be distributed with the source code. Does anyone have an opinion on this? Also, for anyone who wants to check-out this implementation, I can provide more details. Thanks, Vikrant _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

4 3

[Users] UAC
by Kapil Dhawan 26 Dec '05

26 Dec '05

Hi List I am using Cisco 2600 to send calls to OpenSER but 2600 is not sending the username in from field, so i used UAC module. Issues. 1. I need to make changes in openser.cfg and restart service, do we have any other alternative. 2. My calls are getting disconnected after a random time interval say couple of minutes. KD _________________________________________________________________ One and only Ash. Find out all about her. Only on MSN Search http://server1.msn.co.in/profile/aishwarya.asp

2 1

[Users] Context of flags - flag visibility - in openser
by Joachim Fabini 23 Dec '05

23 Dec '05

Hi, Again a question regarding the OpenSER documentation, specifically the usage of flags: What is the context and visibility scope of flags in openser? What is their lifetime? Maybe some piece of code to clarify things: route { if (method=="REGISTER") { setflag(10); # do some processing } if ((method=="INVITE")||(method=="BYE")) { setflag(11); } # do some more processing t_relay(); } onreply_route() { if (isflagset(10)) { # received a reply on register, process it } if (isflagset(11)) { # reply on invite or bye } } a) Can we expect the flags 10/11 to be identical for route and onreply_route - i.e. in the example above, are the flags in status "set" when processing the onreply_route? I.e. isflagset(10) returns true if we process the 200 OK in reply to a REGISTER? b) Do we need to explicitely reset the flag or is it automatically cleared? Specifically: If an INVITE and the matching OK are processed, is it required to reset the flag afterwards in onreply_route? I.e., if after the 200OK to INVITE a 200 OK reply to a SUBSCRIBE is processed by onreply_route and we did _not_ reset the flag 11, does isflagset(11) return true or false on the 200 OK for SUBSCRIBE? Finally another question: must onreply_route be explicitely triggered (like for onreply_route[1] using t_on_reply("1")) or is the default onreply_route anyhow executed automatically on any stateful reply? Thanks in advance, --Joachim

3 2

[Users] Timer for unaccessible gateway
by Taras Bendik 23 Dec '05

23 Dec '05

Hi I have a problem with gateways. I use this two timers: modparam("tm", "fr_timer",15) modparam("tm", "fr_inv_timer",30) However my problem is , when some carrier not respond at all for first 5 seconds, which timer is for that? Regards, Taras

3 6

[Users] Happy Holidays
by Bogdan-Andrei Iancu 23 Dec '05

23 Dec '05

Dear OpenSER community, On behalf of OpenSER team, I would like to wish everybody a Merry Christmas and a Happy New Year. I hope that 2006 will be a fruitful year for both the users and the project itself ;). Best regards and wishes, Bogdan

1 0

[Users] OpenSER 1.1
by Douglas Garstang 22 Dec '05

22 Dec '05

How can I download 1.1 of OpenSER? The docs at http://openser.org/index.php#download don't tell how to do this. I tried: cvs -z3 checkout -r rel_1_1_0 sip-server but it didn't work. Then I tried to get the latest cvs and it fails to compile. It fails at the same point it failed at last week. gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -Werror -DNAME='"openser"' -DVERSION='"1.1.0-dev4 "' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c openserunix.c -o openserunix.o cc1: warnings being treated as errors openserunix.c: In function âEUR~mainâEUR(tm): openserunix.c:93: warning: pointer targets in passing argument 6 of âEUR~recvfromâEUR(tm) differ in signedness make[1]: *** [openserunix.o] Error 1 make[1]: Leaving directory `/root/sip-server/utils/openserunix' make: *** [utils/openserunix/openserunix] Error 2 Thanks, Doug.

2 1

[Users] USRLOC - contact matching algorithm
by Bogdan-Andrei Iancu 22 Dec '05

22 Dec '05

Hi, Based on the latest discussions about inefficiency if the former contact matching alg, in the context of NAT traversal, I just committed into USRLOC support for configurable multi-algorithm for contact matching. For what algorithm are available, their description and how to select them, please refer to the online docs: http://www.openser.org/docs/modules/1.1.x/usrloc.html#CONTACT-MATCHING-ALGS I hope it will solve most of the problems... ;) regards, bogdan

1 0

[Users] RE: [Devel] dispatcher has failover support
by Douglas Garstang 22 Dec '05

22 Dec '05

Can someone *PLEASE* post an example of how this works with failure_route? I can't get the ds_next_dst() command to work, probably because I don't know HOW! Doug. -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Sunday, December 18, 2005 3:07 AM To: devel(a)openser.org; users(a)openser.org Subject: [Devel] dispatcher has failover support Hello, the dispatcher module (kind of fast load balancing support) has now support for failover. Along with selected destination, the rest of addresses in the destination set are stored in AVP list. If the selected destination fails to manage the request, then the next address can be used from failure_route. Some destinations can be dynamically marked as inactive (e.g., based on reply code) so the are not going to be used until the admin enables them again, via fifo command. For more info, please see the readme file: http://openser.org/docs/modules/1.1.x/dispatcher.html Cheers, Daniel _______________________________________________ Devel mailing list Devel(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/devel

1 0

Re: [Users] AVPops question
by Bogdan-Andrei Iancu 22 Dec '05

22 Dec '05

Hi Joachim, why do you need a variable key name? wouldn't this work for you? avp_printf("i:10", "$hdr(call-id)-$Ts"); avp_pushto("$Myheader","i:10"); regards, bogdan Joachim Fabini wrote: >Hi, > >Is there a way or workaround to generate AVPs with header >values as AVP name? It seems to me that the current AVP >concept defines the AVPs at OpenSER startup, so >header-value-based names are not feasible. > >What I need is the following: Store/retrieve a (key,value) >pair where the key is the current message's call-id and >the value some combination of header values and pseudo- >variables. > >e.g. use the ops >avp_printf("$hdr(call-id)", "$hdr(call-id)-$Ts"); >avp_pushto("$Myheader","$hdr(call-id)"); > >in order to append the following header to the SIP message: >Myheader: 23459(a)10.0.0.1-4235627623 > >Any idea how to do this in OpenSER? > >thanks in advance >--Joachim > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

2 2

[Users] OpenSer questions
by Tim Barrett 22 Dec '05

22 Dec '05

I would greatly appreciate knowing whether the following is possible with openser and rtpproxy/nathelper. 1) I have a linksys router, NAT enabled with a SIP UA 2) I have an Openser/nathelper/rtpproxy server in the public internet space (redhat as 3) 3) I have a SIP UA connected in a non nated environment Running openser and rtpproxy with the enclosed configuration results in call setup but no audio forwarded by openser from the NATED SIP UA to the non nated SIP UA. The non nated SIP UA initiates the call. Stopping the rtpproxy results in receipt of audio from the nated SIP UA but no audio is received from the non nated SIP UA. Having spent numerous hours on this, I wonder if this is even possible. The common questions answered below: 1) rtpproxy source was downloaded from the berilos CVS server and built locally 2) the openser version is 1.0.0 downloaded the rpm from openser.org Openser.cfg ------------------------ # # $Id: openser.cfg,v 1.4 2005/10/27 19:58:01 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" #--NatHelper loadmodule "/usr/lib/openser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/openser/modules/auth.so" #loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- NATHelper -- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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(nat_uac_test("3")) { if (method == "REGISTER" || !search("^Record-Route:")) { log("LOG: Someone trying to register from private IP\n"); fix_nated_contact(); if (method == "INVITE") { log("LOG: Fix nated sdp - INVITE \n"); fix_nated_sdp("1"); # record_route(); # force_rtp_proxy(); # t_on_reply("1"); # t_relay(); }; force_rport(); setflag(6); }; }; if (!method=="REGISTER") { record_route(); log("LOG: Someone trying to register from private IP\n"); # fix_nated_contact(); }; # 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"); t_relay(); setflag(1); return; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # 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("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; log("LOG: Caught NAT Register sFlag 6 forcing rtp proxy\n"); setflag(6); # force_rport(); save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } # Was #route[1] { # # send it out now; use stateful forwarding as it works reliably # # even for UDP2TCP # if (!t_relay()) { # sl_reply_error(); # }; # exit; #} route[1] { # !! Nathelper #if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ # sl_send_reply("479", "no forwarding to private IP"); # return; # }; if (isflagset(6)) { log("LOG: Caught NAT Flag 6 forcing rtp proxy\n"); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } # ! Nathelper onreply_route[1] { # if (isflagset(2) && status =~"(183)|2[0-9][0-9]") { if (status =~"(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

2 1

[Users] help: using mediaproxy
by unplug 22 Dec '05

22 Dec '05

Hi, I am using the example "gw-pstn" from Getting started. I followed the guide to setup mediaproxy for handle NAT function. However, I can't make call between Internal IPphone to PSTN. Anyone can help me to analyze why it doesn't work. After using mediaproxy, I found that some data are stored in the table "Location" compared not using mediaproxy. The field "contact" contains "sip:number@internalIP:port" and the field "received" contains "sip:externalIP:port". Is it correct after using mediaproxy? I have a look for the log from ngrep and some extra parameters "rport and receieved" in the header. When I try to make a call from internal IP phone to PSTN, the error "500 FQDN in SDP Media cannot be resolved.". I think it is still a NAT problem and it seems to use the internal IP instead of external IP. As the same time, Cisco shows (no route to destination (3)). When I try to make a call from PSTN to internal IP phone, the error " Too Many Hops", which is the error message in the openser.cfg file, shown. I do think it is the same reason as the above case. Below are the log from ngrep during making a call. openser.cfg and mediaproxy.ini are also listed below. Please help to figure it out the problem. Many thanks! >>>>>>>>>>>>>-------start of the log ( make a call from internal IP phone to PSTN) --------- interface: eth1 (INTERFACE_IP/255.255.255.240) filter: (ip) and ( port 5060 ) # U NAT_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 INVITE sip:934163634@owthkems01.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bKwgnpocaCevAIebRx. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Contact: <sip:36418497@10.0.0.63:5060>. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 290. . v=0. o=36418497 63608336 94989291 IN IP4 10.0.0.63. s=SIP CALL. c=IN IP4 10.0.0.63. t=0 0. m=audio 6000 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bKwgnpocaCevAIebRx;rport=5060;received=NAT_PUBLIC_IP. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. CSeq: 1 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 OPENSER_PUBLIC_IP:5060 "Noisy feedback tells: pid=30084 req_src_ip=NAT_PUBLIC_IP req_src_port=5060 in_uri=sip:934163634@owthkems01.mydomain.com out_uri=sip:934163634@CISCO_PUBLIC_IP via_cnt==1". . # U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 INVITE sip:934163634@CISCO_PUBLIC_IP SIP/2.0. Record-Route: <sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on>. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0. Via: SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Contact: <sip:36418497@NAT_PUBLIC_IP:5060>. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 315. . v=0. o=36418497 63608336 94989291 IN IP4 10.0.0.63. s=SIP CALL. c=IN IP4 OPENSER_PUBLIC_IPOPENSER_PUBLIC_IP. #it is very strange as it displayed 2 IPs together! t=0 0. m=audio 3502435024 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U CISCO_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0,SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:17 GMT. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U CISCO_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0,SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:17 GMT. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 5664. Allow-Events: telephone-event. Contact: <sip:934163634@CISCO_PUBLIC_IP:5060>. Record-Route: <sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 184. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6412 5565 IN IP4 CISCO_PUBLIC_IP. s=SIP Call. c=IN IP4 CISCO_PUBLIC_IP. t=0 0. m=audio 16398 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:17 GMT. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 5664. Allow-Events: telephone-event. Contact: <sip:934163634@CISCO_PUBLIC_IP:5060>. Record-Route: <sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 184. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6412 5565 IN IP4 CISCO_PUBLIC_IP. s=SIP Call. c=IN IP4 OPENSER_PUBLIC_IP. t=0 0. m=audio 35024 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U NAT_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 PRACK sip:934163634@CISCO_PUBLIC_IP:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bK6CCN6qgnaIMWA8ww. Route: <sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on>. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Contact: <sip:36418497@10.0.0.63:5060>. CSeq: 2 PRACK. RAck: 5664 1 INVITE. Content-Length: 0. . # U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 PRACK sip:934163634@CISCO_PUBLIC_IP:5060 SIP/2.0. Record-Route: <sip:OPENSER_PUBLIC_IP;ftag=EInGXmoIRNHvq87x;lr=on>. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKc09e.ba203413.0. Via: SIP/2.0/UDP 10.0.0.63:5060;received=NAT_PUBLIC_IP;branch=z9hG4bK6CCN6qgnaIMWA8ww. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Contact: <sip:36418497@10.0.0.63:5060>. CSeq: 2 PRACK. RAck: 5664 1 INVITE. Content-Length: 0. . # U CISCO_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKc09e.ba203413.0,SIP/2.0/UDP 10.0.0.63:5060;received=NAT_PUBLIC_IP;branch=z9hG4bK6CCN6qgnaIMWA8ww. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 PRACK. Content-Length: 0. . # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.63:5060;received=NAT_PUBLIC_IP;branch=z9hG4bK6CCN6qgnaIMWA8ww. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 PRACK. Content-Length: 0. . # U NAT_PUBLIC_IP:15061 -> OPENSER_PUBLIC_IP:5060 ................ # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 .... # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:15061 .... # U OPENSER_PUBLIC_IP:5060 -> 210.22.7.147:2185 .... # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:15060 .... # U CISCO_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 SIP/2.0 500 FQDN Query in SDP Media cannot be resolved. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0,SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 ACK sip:934163634@CISCO_PUBLIC_IP SIP/2.0. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. CSeq: 1 ACK. User-Agent: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. . # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 500 FQDN Query in SDP Media cannot be resolved. Via: SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U NAT_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 ACK sip:934163634@owthkems01.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bKwgnpocaCevAIebRx. Route: <sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on>. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Contact: <sip:36418497@10.0.0.63:5060>. CSeq: 1 ACK. Content-Length: 0. . # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 500 FQDN Query in SDP Media cannot be resolved. Via: SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U NAT_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 ACK sip:934163634@owthkems01.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bKwgnpocaCevAIebRx. Route: <sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on>. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" <sip:36418497@owthkems01.mydomain.com>;tag=EInGXmoIRNHvq87x. To: "934163634" <sip:934163634@owthkems01.mydomain.com>;tag=2D902FF8-1CD1. Call-ID: xZyhO2TPQ3VEaazQ(a)10.0.0.63. Contact: <sip:36418497@10.0.0.63:5060>. CSeq: 1 ACK. Content-Length: 0. . exit 21 received, 0 dropped <<<<<<<<<<<<<<<<<<<-------end of the log ( make a call from internal IP phone to PSTN) --------- >>>>>>>>>>>>>>>>>>>-------start of the log (make a call from PSTN to internal IP phone) ----------- interface: eth1 (203.193.46.224/255.255.255.240) filter: (ip) and ( port 5060 ) # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:15061 .... # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:15060 .... # U CISCO_PUBLIC_IP:56976 -> OPENSER_PUBLIC_IP:5060 INVITE sip:36418498@OPENSER_PUBLIC_IP:5060 SIP/2.0. Via: SIP/2.0/UDP CISCO_PUBLIC_IP:5060. From: <sip:34163634@CISCO_PUBLIC_IP>;tag=2DA393B0-1816. To: <sip:36418498@OPENSER_PUBLIC_IP>. Date: Sat, 17 Dec 2005 06:33:27 GMT. Call-ID: DDE5838A-6DFD11DA-82FEA518-578F6E21@CISCO_PUBLIC_IP. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 3722686073-1845301722-2197529880-1469017633. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 15. Remote-Party-ID: <sip:34163634@CISCO_PUBLIC_IP>;party=calling;screen=yes;privacy=off. Timestamp: 1134801207. Contact: <sip:34163634@CISCO_PUBLIC_IP:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 304. . v=0. o=CiscoSystemsSIP-GW-UserAgent 366 9285 IN IP4 CISCO_PUBLIC_IP. s=SIP Call. c=IN IP4 CISCO_PUBLIC_IP. t=0 0. m=audio 18044 RTP/AVP 3 18 4 2 15. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=yes. a=rtpmap:2 G726-32/8000. a=rtpmap:15 G728/8000. # U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP CISCO_PUBLIC_IP:5060. From: <sip:34163634@CISCO_PUBLIC_IP>;tag=2DA393B0-1816. To: <sip:36418498@OPENSER_PUBLIC_IP>. Call-ID: DDE5838A-6DFD11DA-82FEA518-578F6E21@CISCO_PUBLIC_IP. CSeq: 101 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 OPENSER_PUBLIC_IP:5060 "Noisy feedback tells: pid=30091 req_src_ip=CISCO_PUBLIC_IP req_src_port=56976 in_uri=sip:36418498@OPENSER_PUBLIC_IP:5060 out_uri=sip:36418498@OPENSER_PUBLIC_IP:5060 via_cnt==1". . # U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 SIP/2.0 483 Too Many Hops. Via: SIP/2.0/UDP CISCO_PUBLIC_IP:5060. From: <sip:34163634@CISCO_PUBLIC_IP>;tag=2DA393B0-1816. To: <sip:36418498@OPENSER_PUBLIC_IP>;tag=b5ab8e75af536221ca172f8fc39505eb.bdf6. Call-ID: DDE5838A-6DFD11DA-82FEA518-578F6E21@CISCO_PUBLIC_IP. CSeq: 101 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 OPENSER_PUBLIC_IP:5060 "Noisy feedback tells: pid=30091 req_src_ip=OPENSER_PUBLIC_IP req_src_port=5060 in_uri=sip:36418498@OPENSER_PUBLIC_IP:5060 out_uri=sip:36418498@OPENSER_PUBLIC_IP:5060 via_cnt==16". . # U CISCO_PUBLIC_IP:56976 -> OPENSER_PUBLIC_IP:5060 ACK sip:36418498@OPENSER_PUBLIC_IP:5060 SIP/2.0. Via: SIP/2.0/UDP CISCO_PUBLIC_IP:5060. From: <sip:34163634@CISCO_PUBLIC_IP>;tag=2DA393B0-1816. To: <sip:36418498@OPENSER_PUBLIC_IP>;tag=b5ab8e75af536221ca172f8fc39505eb.bdf6. Date: Sat, 17 Dec 2005 06:33:27 GMT. Call-ID: DDE5838A-6DFD11DA-82FEA518-578F6E21@CISCO_PUBLIC_IP. Max-Forwards: 15. Content-Length: 0. CSeq: 101 ACK. . exit 7 received, 0 dropped <<<<<<<<<<<<-------start of the log (make a call from PSTN to internal IP phone) ----------- >>>>>>>>>---------start of the openser.cfg ---------------------- # $Id: gw-pstn.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=OPENSER_IP port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:[hidden]/openser" log_facility=LOG_LOCAL0 alias=owthkems01.mydomain.com loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" #loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://openser:[hidden]/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) #The default INVITE timer will be allowed 27 seconds to connect. #If we are going to call a PSTN destination then we will set the 'inv_timeout' #AVP to a higher value in order to allow additional timer for the call to #connect. This is show in the PSTN route handler. modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") #modparam("permissions", "db_mode", 1) #modparam("permissions", "trusted_table", "trusted") route { #xlog("L_INFO","SIP Request: method [$rm] from [$si] [$sp] to [$Ri] [$Rp]\n"); # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("OPENSER_IP:5060;nat=yes"); #} else if (method=="ACK") { #zzz # record_route(); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # is_uri_host_local will determine if the domain portion of the # Request URI is a domain that our SIP proxy is responsible for. # ----------------------------------------------------------------- if (!is_uri_host_local()) { #xlog("L_INFO","SIP route4: line 121\n"); #if (is_from_local() || allow_trusted()) { route(4); route(1); #} else { #sl_send_reply("403", "Forbidden"); #}; return; }; if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- # when dealing with NATed clients we must correctly handle reponse # messages that may be heading back to the client. These # response messages are accessible in SER by using a reply_route # block t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { xlog("L_INFO","SIP : line 212\n"); # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- #if (!allow_trusted()) { #if (!proxy_authorize("","subscriber")) { #proxy_challenge("","0"); #return; #} else if (!check_from()) { #sl_send_reply("403", "Use From=ID"); #return; #}; #consume_credentials(); #}; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; #if (uri=~"^sip:1[0-9]{10}@") { # strip(1); #}; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; #if (uri=~"^sip:011[0-9]*@") { # International PSTN #route(4); #route(5); #return; #}; if (!lookup("location")) { if (uri=~"^sip:9[0-9]{8}@") { # Domestic PSTN route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- #xlog("L_INFO","SIP route4: line 274\n"); if (isflagset(6) || isflagset(7)) { xlog("L_INFO","SIP : line 275 use_media_proxy\n"); use_media_proxy(); }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("203.193.46.242"); # INSERT YOUR PSTN GATEWAY IP ADDRESS #xlog("L_INFO","SIP route5: line 287\n"); avp_write("i:45", "inv_timeout"); route(4); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } <<<<<<<<<<<<--------------end of openser.cfg------------------ >>>>>>>>>>>------------start of mediaproxy.ini---------------- ; ; Configuration file for MediaProxy ; [Dispatcher] ; ; Section for configuring the proxy dispatcher ; ; The following options are available here: ; ; start Boolean value that specifies if to start the dispatcher. ; Default value: Yes ; ; socket Path to the UNIX socket where the dispatcher receives ; commands from SER. This should match the value for ; mediaproxy_socket in ser.cfg ; Default value: /var/run/proxydispatcher.sock ; ; group Put the socket in this group and make it group writable. ; Default value: ser ; ; defaultProxy Default mediaproxy to use in case the From/To domains ; involved in the call don't define any. ; Valid values for this are: ; ; - None ; don't use any default proxies. domains without ; mediaproxy SRV records won't work ; - /path/to/unix/socket ; use a single MediaProxy server identified by the given ; UNIX socket path ; - IP_or_hostname[:port] ; use a single MediaProxy server identified by its network ; address. The network address consists of an IP address ; or a hostname and an optional port number separated by ; a double colon. If port is missing 25060 will be assumed. ; Examples: ; 10.0.0.1 (connect to 10.0.0.1 on port 25060) ; 10.0.0.1:90 (connect to 10.0.0.1 on port 90) ; mp1.mydomain.com ; mp1.mydomain.com:7000 ; - domain://domain_name ; Use all MediaProxies defined by domain_name, honoring ; their priority and weight to create a cluster of proxies ; with fallback and load balancing capabilities. ; ; Default value: /var/run/mediaproxy.sock ; ;start = yes ;socket = /var/run/proxydispatcher.sock ;group = ser ;defaultProxy = /var/run/mediaproxy.sock [MediaProxy] ; ; Section for configuring the MediaProxy server ; ; The following options are available here: ; ; start Boolean value that specifies if to start the RTP proxy server. ; Default value: Yes ; ; socket Path to the UNIX socket where MediaProxy receives ; commands from the dispatcher or SER. ; Default value: /var/run/mediaproxy.sock ; ; group Put the socket in this group and make it group writable. ; Default value: ser ; ; listen Network address where MediaProxy receives commands from ; a remote dispatcher. ; Valid values for this are: ; ; - None ; don't listen for network connections at all ; - address[:port] ; listen on the specified address and port ; address can be an IP a hostname or the keyword Any ; (in which case it will listen on 0.0.0.0). If address is ; a hostname, that should map in DNS to an IP address ; present on the machine through an A record. ; If port is missing assume 25060. ; ; Default value: None ; ; allow List of addresses that are allowed to connect to this ; MediaProxy server and send commands. ; They are specified as a comma separated list of entries, with ; each entry being specified in the CIDR network/mask notation ; (ex. 10.0.0.0/8) ; ; In addition simple IP addresses or hostnames are allowed, in ; which case the mask is considered to be 32. ; ; In addition to network ranges/addresses 2 keywords can be used ; for this option: ; None to specify that none is allowed to connect (not very ; useful but this is the default for security reasons) ; Any to specify that anyone is allowed to connect ; (dangerous!) ; ; Example: allow = 10.0.0.0/24, home-pc.mydomain.com, 1.2.3.4 ; ; Default value: None ; ; proxyIP IP address to use to talk to the phones. If not specified, the ; first found will be used. However first found usually means ; first defined in /etc/hosts which may not be what you want. ; If you find that the address that's automatically selected is ; not the one you want, you can specify the right one using this ; option. The address must be one that's present on one of the ; host's interfaces. ; ; portRange The range of ports to use for proxying the rtp streams. ; This option is specified as minport:maxport with minport and ; maxport being even numbers in the range 1024-65536 ; Default value: 35000:65000 ; ; TOS Unless you know what TOS means, leave this option alone. ; The TOS value can be specified either as a decimal number or ; as a hex number in the 0xnn format. ; Default value: 0xb8 ; ; idleTimeout Expire idle sessions after this much time. ; Default 60 seconds ; ; holdTimeout Expire calls on hold after this much time. ; Default value is 3600 seconds ; ; forceClose Forcibly close a RTP session after this many seconds even if ; it's still active. If forceClose is 0, then a session is never ; closed no matter how long it lasts. ; Default value: 0 ; ;start = yes ;socket = /var/run/mediaproxy.sock ;group = ser ;listen = None ;allow = None ;proxyIP = 10.0.0.1 ;portRange = 35000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 accounting = off [Accounting] user = dbuser password = dbpass host = dbhost database = radius table = radacct <<<<<<<<<<<-----------end of mediaproxy.ini-------------

2 3

[Users] High Availability Setup
by samuel 21 Dec '05

21 Dec '05

Hi verybody, WARNING: Long e-mail..... I just started learning how to set up a "classical" configuration with load balancer, DB backend and so on....I was reading several threads of the mailing lists but since I am a begginer I would need some help and clarifications. Apologies if my questions have been solved before. [I read Greger and Granig (and more people, I guess) are working (?) on a document describing what I try to explain in the follofwing lines. I would like to know if you are on it or commercial strategies forced you to leave it...as it seems that HA setups are high confidential ;)] I'll try to draw what I have guessed from the threads and other literature... [SER1] [LB1] [SER2] . [DB (mysql cluster)] [LB2] . [SERn] LBi are the front-end load balancers. SER: do I have to say what are these?? (either openSER/SER. I will use the name SER for simplicity but I refer to both) DB: I placed mysql cluster because it seems to be the most common backend... Let me make an overview... Load Balancer For LB, it seems that dispacther module can be used. (I don´t remember who proposed to dispatch REGISTER with To: and INIVTE with Req-URI, why not using Call-ID?? For distributing the registration of a given user between several registrars?). Dispatcher module does not have failover in case SERi fails so somebody propose to use lcr. Using lcr module provides failover but does not guarantee to balance to the right instance of SER so I think the only option is to use dispatcher. Alternatives: Please correct me if I am wrong but there isn´t yet an LVR implementation being Call-ID aware.... DNS SRV records...lack of implementation in end-points and not NAT-friendly.. In order to provide high availability, I tried linux-ha between the 2 LBi for simplicity although it has some wasted resources. Is it worth to give a try to other IP failover aplications (anycast, quagga/bgp, vrrpd)? Anyone with more experience can comment their use?? SIP proxy SER....just a matter of taste what you want to use, SER or openSER.... I read about the cacheless usrloc patch but I could not find it on CVS...has it been integrated into the head branch??? has it been deprecated because of ...(please fill in the reason)? DB Backend For the DB backend...I am going to comment mysql clustering...I am using now 4.1.11 because it was already installed. Is it recommended to upgrade?? I tried setting the management node, a couple of data nodes, and I was trying to set a linux-ha configuration with ip take over for the mysql api node (is it possible??) but unsuccessfully: When the preferred mysql api node falls down, SER detects the connection is closed and all further requests are replied with "Internal Server Error" although the secondary mysql api has overtaken the service IP and ser database is properly populated. Is it possible to make SER reconnecting on mysql failure?? It reconnects to the preferred server once it comes back to live but it is impossible to connect to the secondary.... As you might have guessed I am a total begginer with mySQL... Thank for reading until this line, all kinds of comments are wellcome, Samuel.

3 5

[Users] How to configure openser?
by Lucius Curado 21 Dec '05

21 Dec '05

Hi: Where can I find information about configuring a simple openser which permit communication between two machines at the same network? Looking at the openser.org I could not find any information about it. Thanks for helping. Best regards, shakuhashi.

2 1

Re: [Users] Radius Multi Call Leg Accounting/Billing
by Klaus Darilion 20 Dec '05

20 Dec '05

Hi Lenir! I've never used, thus I do not know how to use it. MAybe some can help us: What should be written into src and dst AVPs? The SIP URI of B and C (call forwarding scenario)? Which radius types will be used for them? regards klaus Lenir wrote: > Hey guys, > > I sent this as a duplicate for eaiser searching for others to find. > > pardon my ignorance...Klaus that was help full, thank you. > Ok, so I know how/which AVPs to populate. But I still don't know when to > populate them. > > Let me elaborate more on this, Im doing radius accounting, so I don't know > when to set the flag in conjunction with when to populate the AVPs. > > Will the avp_write() function trigger a radius star/stop record when the > AVPs are populated? > > Do I use a radius function in conjunction with all of this? > > If anybody has a brief example or a snippet of their config, this would > greatly help. > > > Thanks, > > Lenir > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Thursday, December 15, 2005 5:29 AM > To: Lenir > Cc: users(a)openser.org > Subject: Re: [Users] Multi Call Leg Accounting/Billing > > Hi Lenir! > > You should read the emails on the mailing list (or the archive). > Read the thread initiated from Helge on 8.12.2005 > > klaus > > Lenir wrote: > >>Hello guys, >> >> >> >>I read in the documentation for the ACC module that it supports multi >>call-leg accounting via the use of AVPs, which comes in handing when >>doing call forwarding. However, I've googled the way with all the >>keywords that I could think of and I cant find an example of this >>feature. I understand that you have to turn it on using: >> >> >> >>modparam("acc", "multi_leg_enabled", 1) >> >>modparam("acc", "src_leg_avp_id", 110) >> >>modparam("acc", "dst_leg_avp_id", 111) >> >> >> >>But it doesn't show or say when/how to use the AVPs. Can anybody provide >>a sample snippet or their openser.cfg file to demonstrate how it's used? >> >> >> >> >> >>Thanks in advance! >> >> >> >>Lenir >> >> >> >>P.s. perhaps put a sample config in the documentation. The one there >>doesn't show how to use Multi Call-leg billing. >> >> >> >> >> >> >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

2 1

[Users] delay test
by Daniel-Constantin Mierla 20 Dec '05

20 Dec '05

please ignore ...

1 1

[Users] Reply Route (t_on_reply)
by Joachim Fabini 20 Dec '05

20 Dec '05

Hi, Does OpenSER support sequentially processing the same reply by several onreply routes, i.e. route { # general case - have all replies processed by # onreply_route[1] t_on_reply("1"); if (mycondition) { # special case - these replies should be processed # by onreply_route[1] _and_ by onreply_route[2] t_on_reply("2"); } t_relay(); } onreply_route[1] { # append_hf("P-MyExtraInfo: xyz\r\n"); } onreply_route[2] { # do some extra processing here } I did not find any info on this topic in the docs. Imho the ability to use trigger several onreply_routes for sequential processing can help extremly in keeping code clean and readable. Assume a proxy that is required, e.g., to insert a new header field into _any_ reply. In addition, some other replies need some extra handling in the onreply route. Without the capability to trigger multiple onreply-routes this leads to many conditionals in either the main route or in the reply route(s). Thanks in advance, --Joachim

3 4

[Users] Dispatcher vs. LCR?
by Matt Schulte 20 Dec '05

20 Dec '05

Ok, other than the fact that you have pattern matching in lcr, what are the key differences between Dispatcher and LCR? Anyone have any examples or howto's to point me to? Sincerely, Too cheap to by a SLB.. :-)

2 1

RE: [Users] OpenSER and transport selection (SRV and NAPTR)
by Joachim Fabini 19 Dec '05

19 Dec '05

Hi Bogdan, > I meant t_reply() will keep its behaviour as looking into URI for the > destination - but it will incorporate the NAPTR lookup. Great - this answers my question. > to response also to on earlier question, regarding the timing > - I say 3 months are more than enough ;). Excellent. Many thanks for your help (and patience ;) --Joachim > Joachim Fabini wrote: > > >Hi Bogdan, > > > > > > > >>indeed, there was a misunderstanding :) .t_relay() with no > >>param will be kept with the current functionality. > >>Or you suggest to be able to specify only a different proto > >>or port from the RURI? > >> > >> > > > >I did not yet find the primitive which is supposed to > >statefully relay and do the following: > >1. NAPTR query on the transport to use PLUS > >2. _exactly_ what t_relay() does now for the > > transport returned by the NAPTR query. > > > >You say that t_relay() is kept with the current functionality. > >Does this mean no NAPTR, or will t_relay() be extended to use > >NAPTR before SRV/A query? If the latter then everything is OK. > > > >Otherwise we have the alternatives: > > > >t_relay() -> Stateful relaying according to destination > > address using the incoming transport (no NAPTR). > >t_relay_to() -> Stateful relaying to a specific host (you said > > that <host> is mandatory here) using NAPTR to > > determine the transport. > > > > > >To summarize: > >My point is that none of these two primitives covers the case > >when the message is to be relayed to the next hop based only > >on the destination address _and_ using the transport selected > >by the destination proxy (determined via NAPTR query). > > > >Except if either a) t_relay() does NAPTR or b) the host > >parameter in t_relay_to() is optional. > > > >--Joachim > > > > > > > > >

1 0

[Users] OpenSER and transport selection (SRV and NAPTR)
by Joachim Fabini 19 Dec '05

19 Dec '05

Hi, Initially I intended to submit a feature request for OpenSER functionality similar to t_relay_to_udp, t_relay_to_tcp but WITHOUT the need to specify the destination IP address and port. My reasoning was to let the new functions do exactly what t_relay() does but in addition force the DNS SRV query for a specific transport (udp, tcp, or tls). Seeing the roadmap for 1.1 I realized that NAPTR solves part of this problem. Is there already any estimation when NAPTR implementation will be available in OpenSER? I intentionally said "part of the problem" because NAPTR delegates the decision on the transport to the destination DNS. But: our own proxy might also be required to control the selected transport. It seems to me quite flexible to implement a proxy-hosted NAPTR emulation and specify the list of requested protocols, e.g. t_relay_to_transport("tls","tcp",0) specifies that t_relay should firstly generate a SRV DNS query for _sips._tcp.mydomain.org, if this fails to query for _sip._tcp.mydomain.org and then give up - if I don't want UDP. This saves one DNS query (NAPTR) and enables a proxy to control what protocols it uses for contacting the next hop. Finally, I believe that the interface to t_relay_naptr() (or whatever the naptr relaying will be called) should support selection of desired protocols, e.g. t_relay_to_naptr("tls", "tcp", "udp") where entries can be left empty to indicate that this transport is not desired. The argument order might even overrule the NAPTR response transport ranking - finally we connect and thus decide on what transport we prefer... ;) Any comments warmly welcome, best regards --Joachim

4 10

RE: [Users] mysql cluster and openser
by Maria Yndefors 19 Dec '05

19 Dec '05

Hallo again, I still can't get openser mysql module to find correct libmysqlclient. So I have changed my setup I now have debian stable with libc: ii libc6 2.3.2.ds1-22 GNU C Library: Shared libraries and Timezone ii libc6-dev 2.3.2.ds1-22 GNU C Library: Development Libraries and Hea dpkg -i openser-mysql-module_1.0.0.0_i386.deb Selecting previously deselected package openser-mysql-module. (Reading database ... 23597 files and directories currently installed.) Unpacking openser-mysql-module (from openser-mysql-module_1.0.0.0_i386.deb) ... dpkg: dependency problems prevent configuration of openser-mysql-module: openser-mysql-module depends on libmysqlclient10; however: Package libmysqlclient10 is not installed. dpkg: error processing openser-mysql-module (--install): I tried installing SER again to see if I still can get it to work and it worked fine. I installed this version of ser: Get:1 http://apt.sip-router.org stable/main ser 0.9.4-0.1 Unpacking ser-mysql-module (from .../ser-mysql-module_0.9.4-0.1_i386.deb) ... Setting up ser-mysql-module (0.9.4-0.1) ... These are the mysql packages I have installed: ii libmysqlclient12 4.0.24-10sarge1 mysql database client library ii libmysqlclient15 5.0.13rc-1 mysql database client library ii mysql-client-5.0 5.0.13rc-1 mysql database client binaries So openser-mysql-module, needs to use the shared library libmysqlclient12 instead of libmysqlclient10? Any ideas? regards /Maria > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: den 15 december 2005 15:10 > To: Maria Yndefors > Cc: users(a)openser.org > Subject: Re: [Users] mysql cluster and openser > > > indeed, if you have debian unstable, use the corresponding openser > package... > > regards, > bogdan > > Maria Yndefors wrote: > > >Ok, > >I have downloaded the packages from deb-stable directory? So > I guess I have to change to the unstable packages... > >I guess libc is not the same on them. > > > >openser_1.0.0.0_i386.deb > >openser-mysql-module_1.0.0.0_i386.deb > > > >Regards > >/Maria > > > > > > >

2 1

[Users] pike module usage
by Klaus Darilion 19 Dec '05

19 Dec '05

Hi! I have a SIP phone sending a REGISTER request every second. I want to block requests from this SIP phone for some time. I've read the docs but IMO the configuration and usage is quite complicated. Can the pike module be used to block this user? I'm not sure how to configure the pike module to block the REGISTER (1 request per second) but to allow proper dialing (e.g. a call setup may have 4 requests within one second). Does the pike module count all requests received or only those requests for which I execute pike_check_req()? regards klaus

2 2

[Users] Public and private interfaces
by Wilko van der Ploeg 19 Dec '05

19 Dec '05

I've got some problem when I try to forward an incoming call to a PSTN gateway : The call comes in over the public network interface and should be forwarded via the private interface to the gateway (which only has a pvt address), but for some reason, the traffic is sent out over the public interface again and -logically- not able to find the gateway. Any idea where to look ? Wilko

2 1

RE: [Users] Transaction already in Process
by Douglas Garstang 19 Dec '05

19 Dec '05

We thought about sending calls back though OpenSER too, and in Asterisk referencing all dial commands with @proxy, but it really was not clear how things like ACD Queues with agent call backs would work for a start. -----Original Message----- From: Iqbal [mailto:iqbal@gigo.co.uk] Sent: Friday, December 16, 2005 11:26 AM To: Douglas Garstang Cc: Klaus Darilion; users(a)openser.org Subject: Re: [Users] Transaction already in Process jumping in alittle late on this one, but i read that routing the calls back via openser is a pain rather than terminating on asterisk. I had the same dilemma sometime ago when doing virtual pbx setup, but i found it easier to send the call back to ser, and let it do all the registration/routing etc, and just use asterisk for features. would be interested on feedback on this....I may have designed it all wrong after reading this :-) Iqbal Douglas Garstang wrote: >Klaus, > >That's it! > >Doug. > >-----Original Message----- >From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >Sent: Friday, December 16, 2005 12:25 AM >To: Douglas Garstang >Cc: users(a)openser.org >Subject: Re: [Users] Transaction already in Process > > >So how does it look like when Client A calls Client B? Is the signaling >Like in the figure? > > Asterisk > / \ > / \ > Openser \ > / \ > / \ >Client A Client B > > >regards >klaus > > > >Douglas Garstang wrote: > > >>Klaus, >> >>All calls are going through OpenSER from the phones. However, they don't go BACK to OpenSER. Asterisk terminates the call. Putting all calls BACK through OpenSER would be a nightmare. It would make a lot of Asterisk features such as ACD Queues, MeetMe Conferencing etc very hard to manage. I'm not even sure if they would work. >> >>Doug. >> >>-----Original Message----- >>From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >>Sent: Thursday, December 15, 2005 8:23 AM >>To: Douglas Garstang >>Cc: users(a)openser.org >>Subject: Re: [Users] Transaction already in Process >> >> >>I can't explain it, but I do not like the idea replicating with failure >>routes ... >> >>IMO it would be better to modify t_replicate to allow to replicate to >>moultiple instances. >> >>Would i be possible to route all calls via openser? Then you do not need >>the location info in the Asterisks. >> >> >>regards >>klaus >> >>Douglas Garstang wrote: >> >> >> >>>Klaus, >>> >>>Asterisk can't use a database for location/contact information. It also has no native means to transfer registrations between itself and another Asterisk system. It can't even perform a user defined action when a phone registers... doesn't leave us with many options. Right now, using OpenSER to replicate (forward,t_relay,t_replicate whatever) seems to be working ok. Do you have any reason to believe it wouldn't scale up well? >>> >>>Doug. >>> >>>-----Original Message----- >>>From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >>>Sent: Thursday, December 15, 2005 3:23 AM >>>To: Douglas Garstang >>>Cc: users(a)openser.org >>>Subject: Re: [Users] Transaction already in Process >>> >>> >>>Hi Douglas! >>> >>>I still think it is a bad idea to replicate REGISTER to Asterisk 4 >>>times. Where does Asterisk store the location table? In a database? Why >>>not replicate the location data on DB level? >>> >>>regards >>>klaus >>> >>>Douglas Garstang wrote: >>> >>> >>> >>> >>>>Arrgh! >>>> >>>>Now I'm getting "t_newtran: transaction already in process" messages whenever I try to call t_relay() or t_replicate twice, even when the first attempt has FAILED. In the case of t_relay() I have new addresses in the destination set and I still get this error. Why? When you attempt to deliver, and it fails, how can to attempt again without getting those messages? What fundamental piece am I missing? >>>> >>>>How could I do this? Forget the fancy stuff. I don't care about checking for failure anymore. >>>> >>>>t_replicate("192.168.10.7","5060"); >>>>t_replicate("192.168.10.8","5060"); >>>>t_replicate("192.168.10.200","5060"); >>>> >>>>Doug. >>>> >>>>_______________________________________________ >>>>Users mailing list >>>>Users(a)openser.org >>>>http://openser.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> >>> >>> >>> >> >> > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >. > > >

2 1

[Users] dispatcher has failover support
by Daniel-Constantin Mierla 18 Dec '05

18 Dec '05

Hello, the dispatcher module (kind of fast load balancing support) has now support for failover. Along with selected destination, the rest of addresses in the destination set are stored in AVP list. If the selected destination fails to manage the request, then the next address can be used from failure_route. Some destinations can be dynamically marked as inactive (e.g., based on reply code) so the are not going to be used until the admin enables them again, via fifo command. For more info, please see the readme file: http://openser.org/docs/modules/1.1.x/dispatcher.html Cheers, Daniel

1 0

[Users] RE: Muisc on hold.
by news 18 Dec '05

18 Dec '05

Hello, I was just wondering if there is a way to get music on hold to work with OpenSER. Hold currently works however it would be nice to transfer the RTP stream to a media server like SEMS or even Asterisk, instead of giving the end user deathly silence. I have tried to make my own script but I think it might need an OpenSER module? Just wanted to know how/if other users are dealing with this. Thank you. Justin

1 0

RE: [Users] Variables
by Douglas Garstang 17 Dec '05

17 Dec '05

Thanks Tim. Klaus responded today with an email that gave me enough information to finally manipulate an AVP, and print it with xlog(). Wooo! -----Original Message----- From: Tim Klein [mailto:tkpublic@timklein.fastmail.fm] Sent: Fri 12/16/2005 7:55 PM To: users(a)openser.org Cc: Subject: RE: [Users] Variables >I found something somewhere that said something like [si]$avp(name) >or whatever. What the heck does [si] mean??? Was there a colon after the "[si]"? If so, then I'll bet [si] was just a regular expression meaning "either the letter 's' or the letter 'i'". My limited understanding is that every AVP reference starts with either "i:" or "s:", depending on whether you're referencing it by its integer name or its string-format name. (The latter is more human-friendly, but slower.) Tim _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

RE: [Users] Variables
by Douglas Garstang 17 Dec '05

17 Dec '05

Tim, I've read those docs many many times. They still don't make any sense. I can't even find an example of how to print an avp with xlog. I found something somewhere that said something like [si]$avp(name) or whatever. What the heck does [si] mean??? Doug. -----Original Message----- From: Tim Klein [mailto:tkpublic@timklein.fastmail.fm] Sent: Thu 12/15/2005 4:01 PM To: users(a)openser.org Cc: Subject: RE: [Users] Variables >Is there any useful information ANYWHERE about avp's? I mean, don't >take this the wrong way, but I've never had so much trouble >interpreting/finding documentation on a subject. At http://www.openser.org there's a link to this set of tutorials: http://www.voice-system.ro/docs/ The one about the AVPops module is excellent. I hope this set of tutorials will grow over time! Tim _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

2 1

[Serusers] Jabber module...
by sip 16 Dec '05

16 Dec '05

I was attempting to muck with the Jabber module from ser-0.9.4 stable release... but, while the module compiles without a hitch, installing it kills the server. I get pages and pages of: Dec 16 15:12:02 death /usr/local/sbin/ser[3993]: get_connection(): Inherited open database connections, this is not a good idea Messages in the log with the occasional Dec 16 15:12:02 death /usr/local/sbin/ser[4000]: db_init(): Could not create a connection I didn't make any config changes. The database exists and is accessible. All I did was load the module and set the modparams. Is the Jabber module no longer really usable? If not, are there alternatives? I know the new presence stuff uses an Xcap server, but I was kind of hoping to hold off on all that until it became a little less constantly tinkered with. :) N.

1 0

RE: [Users] Transaction already in Process
by Douglas Garstang 16 Dec '05

16 Dec '05

Klaus, That's it! Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Friday, December 16, 2005 12:25 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Transaction already in Process So how does it look like when Client A calls Client B? Is the signaling Like in the figure? Asterisk / \ / \ Openser \ / \ / \ Client A Client B regards klaus Douglas Garstang wrote: > Klaus, > > All calls are going through OpenSER from the phones. However, they don't go BACK to OpenSER. Asterisk terminates the call. Putting all calls BACK through OpenSER would be a nightmare. It would make a lot of Asterisk features such as ACD Queues, MeetMe Conferencing etc very hard to manage. I'm not even sure if they would work. > > Doug. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Thursday, December 15, 2005 8:23 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] Transaction already in Process > > > I can't explain it, but I do not like the idea replicating with failure > routes ... > > IMO it would be better to modify t_replicate to allow to replicate to > moultiple instances. > > Would i be possible to route all calls via openser? Then you do not need > the location info in the Asterisks. > > > regards > klaus > > Douglas Garstang wrote: > >>Klaus, >> >>Asterisk can't use a database for location/contact information. It also has no native means to transfer registrations between itself and another Asterisk system. It can't even perform a user defined action when a phone registers... doesn't leave us with many options. Right now, using OpenSER to replicate (forward,t_relay,t_replicate whatever) seems to be working ok. Do you have any reason to believe it wouldn't scale up well? >> >>Doug. >> >>-----Original Message----- >>From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >>Sent: Thursday, December 15, 2005 3:23 AM >>To: Douglas Garstang >>Cc: users(a)openser.org >>Subject: Re: [Users] Transaction already in Process >> >> >>Hi Douglas! >> >>I still think it is a bad idea to replicate REGISTER to Asterisk 4 >>times. Where does Asterisk store the location table? In a database? Why >>not replicate the location data on DB level? >> >>regards >>klaus >> >>Douglas Garstang wrote: >> >> >>>Arrgh! >>> >>>Now I'm getting "t_newtran: transaction already in process" messages whenever I try to call t_relay() or t_replicate twice, even when the first attempt has FAILED. In the case of t_relay() I have new addresses in the destination set and I still get this error. Why? When you attempt to deliver, and it fails, how can to attempt again without getting those messages? What fundamental piece am I missing? >>> >>>How could I do this? Forget the fancy stuff. I don't care about checking for failure anymore. >>> >>>t_replicate("192.168.10.7","5060"); >>>t_replicate("192.168.10.8","5060"); >>>t_replicate("192.168.10.200","5060"); >>> >>>Doug. >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> > >

2 1

[Serusers] Rtpproxy + Nathelper
by Joel n.solanki 16 Dec '05

16 Dec '05

Hello all, Can anybody provide me working ser.cfg with rtpproxy + nathelper. I have already installed rtpproxy. I have gone through Onsip.org download but config didnt work. Can any body provide me the config. Regards, joel

2 1

Re: [Users] AVPops question
by Klaus Darilion 16 Dec '05

16 Dec '05

I I got it right know, you need the following: avp_printf("s:$hdr(call-id)", "$hdr(call-id)-$Ts"); The AVP ist string based, and the name of the AVP is the Call-id. Guess AVPOPs need to be extended for this. What is the main purpose of this new AVP? Adding as header or storing in the database, so that another proxy can access it? avp_pushto("$Myheader","$hdr(call-id)"); But this would just duplicate the Callid header. regards klaus Joachim Fabini wrote: > Hi, > > Is there a way or workaround to generate AVPs with header > values as AVP name? It seems to me that the current AVP > concept defines the AVPs at OpenSER startup, so > header-value-based names are not feasible. > > What I need is the following: Store/retrieve a (key,value) > pair where the key is the current message's call-id and > the value some combination of header values and pseudo- > variables. > > e.g. use the ops > avp_printf("$hdr(call-id)", "$hdr(call-id)-$Ts"); > avp_pushto("$Myheader","$hdr(call-id)"); > > in order to append the following header to the SIP message: > Myheader: 23459(a)10.0.0.1-4235627623 > > Any idea how to do this in OpenSER? > > thanks in advance > --Joachim > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

2 1

[Serusers] Sipsak 0.9.5 possible bug?
by Marcello Lupo 16 Dec '05

16 Dec '05

Hi, after some hours of fighting trying to use sipsak to register a user on a server i lost and decided to test older version 0.9.2 and it is working. The command: sipsak -U -x 120 -a password -u test -p 12.34.56.78 -C sip:s@11.22.33.44:5060 -s sip:test@domain Didn't worked in 0.9.5 but work perfectly on 0.9.2 . In 0.9.5 nothing was exiting from the server and seems like the sipsak was sending the request to itself. What can be the problem?? Bye, Marcello

2 1

[Serusers] SER+Vocal Integration is failing
by rajesh.rajguru＠wipro.com 16 Dec '05

16 Dec '05

Hi, We are trying to call from SER to Vocal , but call is failing saying 403 forbidden. We are using kphone as user agent . UA1 is registered with SER . UA2 is registered with vocal. When we call to UA2 from UA1 , call is failing saying 403 forbidden. But if we call UA1 from UA2 then it is working properly Has anybody faced this type of problem ? Does vocal requires some config changes ? Any pointers on this will be helpful Regards, Rajesh The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[Users] any howto for using mediaproxy with openser
by unplug 16 Dec '05

16 Dec '05

As title, any where I can find such howto?

2 6

RE: [Users] AVPops question
by Joachim Fabini 16 Dec '05

16 Dec '05

> But the name of the header changes for each transaction/dialog? Yes, that's intended - I do not want to store AVPs in a contact-specific way (otherwise, if I use i:11 from Bogdan's example, the script overwrites previous values on a second call from the same terminal, nonregarding that the first call might not yet be closed) but in a call-specific manner, i.e. the attribute name (string) identifies the call-id that the value stored in the AVP belongs to. Like stated in a previous answer - it's about creating a P-Charging-Vector according to RFC 3455 in the core and inserting it into all forward and reply messages. The AVP name should be created and last exactly for the duration of a SIP transaction. The timestamp is required to guarantee the uniqueness of the value (must not be re-used for a duration of at least one month). A distinct AVP name is required for any call (identified by a distinct call id). --Joachim > Joachim Fabini wrote: > >>Not sure if I understand the problem correctly. Does the > name of the > >>header change? > > > > > > No, it does not change during the lifetime of the AVP. > > The name is set to the call-id and stays constant as > > long as the AVP exists. > > The problem is imho the content of pseudo-variables > > can not be used as the names of AVPs. > > > > --Joachim > > > > > >># AVP 101 with alias > >>modparam("avpops","avp_aliases","myheaderavp=I:101"); > >> > >>avp_printf("$myheaderavp", "$hdr(call-id)-$Ts"); > >>avp_pushto("$Myheader","$myheaderavp"); > >> > >>klaus > >> > >>Joachim Fabini wrote: > >> > >>>Hi, > >>> > >>>Is there a way or workaround to generate AVPs with header > >>>values as AVP name? It seems to me that the current AVP > >>>concept defines the AVPs at OpenSER startup, so > >>>header-value-based names are not feasible. > >>> > >>>What I need is the following: Store/retrieve a (key,value) > >>>pair where the key is the current message's call-id and > >>>the value some combination of header values and pseudo- > >>>variables. > >>> > >>>e.g. use the ops > >>>avp_printf("$hdr(call-id)", "$hdr(call-id)-$Ts"); > >>>avp_pushto("$Myheader","$hdr(call-id)"); > >>> > >>>in order to append the following header to the SIP message: > >>>Myheader: 23459(a)10.0.0.1-4235627623 > >>> > >>>Any idea how to do this in OpenSER? > >>> > >>>thanks in advance > >>>--Joachim > >>> > >>> > >>>_______________________________________________ > >>>Users mailing list > >>>Users(a)openser.org > >>>http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >> > > > > >

1 0

[Users] RE: AVPops question
by jjhab＠netzero.com 16 Dec '05

16 Dec '05

HI HO! Correct me if I'm wrong, but it seems you trying to use the AVPops to capture a time stamp of the very first INVITE of a transaction? To get this same information, I've used the acc module. For billing and user call logging purposes, I needed this info and have used: acc_db_request("CallStart", "acc"); in openser.cfg. This logs a record to the acc table containing the sip_callid, timestamps, and in "sip_status" is the string "CallStart". I then had various scripts parse acc to the format I needed for the end user. Hope this helps, jj ------------------------------ Message: 7 Date: Thu, 15 Dec 2005 13:09:20 +0100 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> Subject: Re: [Users] AVPops question To: Joachim Fabini <Joachim.Fabini(a)tuwien.ac.at> Cc: 'Joachim Fabini' <joachim.fabini(a)gmx.net>, users(a)openser.org Message-ID: <43A15CF0.3070206(a)pernau.at> Content-Type: text/plain; charset=ISO-8859-1; format=flowed But the name of the header changes for each transaction/dialog? klaus Joachim Fabini wrote: >>Not sure if I understand the problem correctly. Does the name of the >>header change? > > > No, it does not change during the lifetime of the AVP. > The name is set to the call-id and stays constant as > long as the AVP exists. > The problem is imho the content of pseudo-variables > can not be used as the names of AVPs. > > --Joachim > > >># AVP 101 with alias >>modparam("avpops","avp_aliases","myheaderavp=I:101"); >> >>avp_printf("$myheaderavp", "$hdr(call-id)-$Ts"); >>avp_pushto("$Myheader","$myheaderavp"); >> >>klaus >> >>Joachim Fabini wrote: >> >>>Hi, >>> >>>Is there a way or workaround to generate AVPs with header >>>values as AVP name? It seems to me that the current AVP >>>concept defines the AVPs at OpenSER startup, so >>>header-value-based names are not feasible. >>> >>>What I need is the following: Store/retrieve a (key,value) >>>pair where the key is the current message's call-id and >>>the value some combination of header values and pseudo- >>>variables. >>> >>>e.g. use the ops >>>avp_printf("$hdr(call-id)", "$hdr(call-id)-$Ts"); >>>avp_pushto("$Myheader","$hdr(call-id)"); >>> >>>in order to append the following header to the SIP message: >>>Myheader: 23459(a)10.0.0.1-4235627623 >>> >>>Any idea how to do this in OpenSER? >>> >>>thanks in advance >>>--Joachim >>> >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> > >

2 1

[Users] to change q-value in real time
by Matteo Piazza 16 Dec '05

16 Dec '05

Hi Is it possibile to change the q-value of a contact with openserctl command? Matteo Piazza

3 4

[Users] LCR module Problem
by Pepe 16 Dec '05

16 Dec '05

Hello, Im configuring Openser with LCR module and Im having an extrange behavior, I have 2 gateways, GW1(preference1) and GW2(preference2), GW1(pref.1) / \ ClientA --> OpenSer --> Client B \ GW2 (pref.2) / When I call from Client A to Client B using GW1, all works fine, its the same when hang up Client B or Client A, but when GW1 fail(I provoke it changing codec) and use failure route (GW 2) then if Client A hang up all works fine, but the problem is when is Client B who hang up, its like a new conversation, GW 2 send BYE to openser and Openser just send "503 Service Not avilable - No gateways" to GW2, but doesnt send nothing to ClientA, any idea ???? Thx in advance this is my basic configuration( I have attached full config file): route{ if (!method=="REGISTER") { record_route(); }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if (method=="INVITE") { if (!lookup("location")) { if (!load_gws()) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); return; }; }; if (!next_gw()) { sl_send_reply("503", "Service not available - No gateways"); return; }; }; }; }; t_on_failure("1"); route(1); } route[1] { if (!t_relay()) { sl_send_reply("666","t_relay fallido"); return; }; exit; } failure_route[1] { if (!next_gw()) { t_reply("503", "Service not available - No gateways 1"); return; }; if (!t_relay()) { t_reply("666","t_relay fallido"); return; }; }

2 1

RE: [Users] Variables
by Douglas Garstang 16 Dec '05

16 Dec '05

Is there any useful information ANYWHERE about avp's? I mean, don't take this the wrong way, but I've never had so much trouble interpreting/finding documentation on a subject. I can't even find out how to print the value of an avp with xlog. Is this ACTUALLY DOCUMENTED anywhere? I finally found an example somewhere from a newsgroup posting and used: xlog ("L_INFO","$avp($foo)"); but openser complains with: Dec 15 13:21:01 bil-pdev-3 openser[24718]: ERROR:xl_parse_name: unknow avp alias"foo" Dec 15 13:21:01 bil-pdev-3 openser[24718]: XLOG:xdbg_fixup: ERROR: wrong format[$avp($foo)]! -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Thursday, December 15, 2005 3:43 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Variables Hi Doug, use avp_printf() or avp_subst() to process the content of an avp - this function covers the assigning operation. regards, bogdan Douglas Garstang wrote: >All, > >I can't find any documentation of OpenSER use of variables. I'm sending a lot of info to syslog with xlog(), and with multiple threads (processes?) and multiple calls running, all the output is out of sequence. As a result, I'd like to prefix the debug output with the call-id. The call-id however is very long and I'd like to cut it down and only display a portion of it. > >How can I extract a string from a SIP message, cut it up, extract the portion I want, and print that portion? OpenSER doesn't seem to have any concept of assigning variables by: > >variable = value > >Thanks, >Doug. > > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

3 2

RE: [Users] Transaction already in Process
by Douglas Garstang 16 Dec '05

16 Dec '05

Klaus, All calls are going through OpenSER from the phones. However, they don't go BACK to OpenSER. Asterisk terminates the call. Putting all calls BACK through OpenSER would be a nightmare. It would make a lot of Asterisk features such as ACD Queues, MeetMe Conferencing etc very hard to manage. I'm not even sure if they would work. Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Thursday, December 15, 2005 8:23 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Transaction already in Process I can't explain it, but I do not like the idea replicating with failure routes ... IMO it would be better to modify t_replicate to allow to replicate to moultiple instances. Would i be possible to route all calls via openser? Then you do not need the location info in the Asterisks. regards klaus Douglas Garstang wrote: > Klaus, > > Asterisk can't use a database for location/contact information. It also has no native means to transfer registrations between itself and another Asterisk system. It can't even perform a user defined action when a phone registers... doesn't leave us with many options. Right now, using OpenSER to replicate (forward,t_relay,t_replicate whatever) seems to be working ok. Do you have any reason to believe it wouldn't scale up well? > > Doug. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Thursday, December 15, 2005 3:23 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] Transaction already in Process > > > Hi Douglas! > > I still think it is a bad idea to replicate REGISTER to Asterisk 4 > times. Where does Asterisk store the location table? In a database? Why > not replicate the location data on DB level? > > regards > klaus > > Douglas Garstang wrote: > >>Arrgh! >> >>Now I'm getting "t_newtran: transaction already in process" messages whenever I try to call t_relay() or t_replicate twice, even when the first attempt has FAILED. In the case of t_relay() I have new addresses in the destination set and I still get this error. Why? When you attempt to deliver, and it fails, how can to attempt again without getting those messages? What fundamental piece am I missing? >> >>How could I do this? Forget the fancy stuff. I don't care about checking for failure anymore. >> >>t_replicate("192.168.10.7","5060"); >>t_replicate("192.168.10.8","5060"); >>t_replicate("192.168.10.200","5060"); >> >>Doug. >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

2 1

RE: [Users] Variables
by Douglas Garstang 15 Dec '05

15 Dec '05

So... using an example I found, if I have something like this... avp_printf("i:20", "This is a $rm request with the call-id $hdr(call-id)"); how do I print the value "i:20"? or whatever the heck that means....? -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Thursday, December 15, 2005 3:43 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Variables Hi Doug, use avp_printf() or avp_subst() to process the content of an avp - this function covers the assigning operation. regards, bogdan Douglas Garstang wrote: >All, > >I can't find any documentation of OpenSER use of variables. I'm sending a lot of info to syslog with xlog(), and with multiple threads (processes?) and multiple calls running, all the output is out of sequence. As a result, I'd like to prefix the debug output with the call-id. The call-id however is very long and I'd like to cut it down and only display a portion of it. > >How can I extract a string from a SIP message, cut it up, extract the portion I want, and print that portion? OpenSER doesn't seem to have any concept of assigning variables by: > >variable = value > >Thanks, >Doug. > > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Users] LCR problem, last one
by Pepe 15 Dec '05

15 Dec '05

Sorry again, I have attached both, file config and body message because it cant be readed from web, somehow the body is missing. Regards

1 0

[Users] LCR Problem again
by Pepe 15 Dec '05

15 Dec '05

Sorry, dont know why but the text is missing, here we go again: Hello, Im configuring Openser with LCR module and Im having an extrange behavior, I have 2 gateways, GW1(preference1) and GW2(preference2), GW1(pref.1) / \ ClientA --> OpenSer --> Client B \ GW2 (pref.2) / When I call from Client A to Client B using GW1, all works fine, its the same when hang up Client B or Client A, but when GW1 fail(I provoke it changing codec) and use failure route (GW 2) then if Client A hang up all works fine, but the problem is when is Client B who hang up, its like a new conversation, GW 2 send BYE to openser and Openser just send "503 Service Not avilable - No gateways" to GW2, but doesnt send nothing to ClientA, any idea ???? Thx in advance this is my basic configuration( I have attached full config file): route{ if (!method=="REGISTER") { record_route(); }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if (method=="INVITE") { if (!lookup("location")) { if (!load_gws()) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); return; }; }; if (!next_gw()) { sl_send_reply("503", "Service not available - No gateways"); return; }; }; }; }; t_on_failure("1"); route(1); } route[1] { if (!t_relay()) { sl_send_reply("666","t_relay fallido"); return; }; exit; } failure_route[1] { if (!next_gw()) { t_reply("503", "Service not available - No gateways 1"); return; }; if (!t_relay()) { t_reply("666","t_relay fallido"); return; }; }

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[Users] LCR Problem again
by Pepe 15 Dec '05

15 Dec '05

Sorry, dont know why but the text is missing, here we go again: Hello, Im configuring Openser with LCR module and Im having an extrange behavior, I have 2 gateways, GW1(preference1) and GW2(preference2), GW1(pref.1) / \ ClientA --> OpenSer --> Client B \ GW2 (pref.2) / When I call from Client A to Client B using GW1, all works fine, its the same when hang up Client B or Client A, but when GW1 fail(I provoke it changing codec) and use failure route (GW 2) then if Client A hang up all works fine, but the problem is when is Client B who hang up, its like a new conversation, GW 2 send BYE to openser and Openser just send "503 Service Not avilable - No gateways" to GW2, but doesnt send nothing to ClientA, any idea ???? Thx in advance this is my basic configuration( I have attached full config file): route{ if (!method=="REGISTER") { record_route(); }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if (method=="INVITE") { if (!lookup("location")) { if (!load_gws()) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); return; }; }; if (!next_gw()) { sl_send_reply("503", "Service not available - No gateways"); return; }; }; }; }; t_on_failure("1"); route(1); } route[1] { if (!t_relay()) { sl_send_reply("666","t_relay fallido"); return; }; exit; } failure_route[1] { if (!next_gw()) { t_reply("503", "Service not available - No gateways 1"); return; }; if (!t_relay()) { t_reply("666","t_relay fallido"); return; }; }

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Re: [Users] Multi Call Leg Accounting/Billing
by Klaus Darilion 15 Dec '05

15 Dec '05

Hi Lenir! You should read the emails on the mailing list (or the archive). Read the thread initiated from Helge on 8.12.2005 klaus Lenir wrote: > Hello guys, > > > > I read in the documentation for the ACC module that it supports multi > call-leg accounting via the use of AVPs, which comes in handing when > doing call forwarding. However, I’ve googled the way with all the > keywords that I could think of and I cant find an example of this > feature. I understand that you have to turn it on using: > > > > modparam("acc", "multi_leg_enabled", 1) > > modparam("acc", "src_leg_avp_id", 110) > > modparam("acc", "dst_leg_avp_id", 111) > > > > But it doesn’t show or say when/how to use the AVPs. Can anybody provide > a sample snippet or their openser.cfg file to demonstrate how it’s used? > > > > > > Thanks in advance! > > > > Lenir > > > > P.s. perhaps put a sample config in the documentation. The one there > doesn’t show how to use Multi Call-leg billing. > > > > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] RV: LCR module Problem
by Pepe 15 Dec '05

15 Dec '05

Hello, Im configuring Openser with LCR module and Im having an extrange behavior, I have 2 gateways, GW1(preference1) and GW2(preference2), GW1(pref.1) / \ ClientA --> OpenSer --> Client B \ GW2 (pref.2) / When I call from Client A to Client B using GW1, all works fine, its the same when hang up Client B or Client A, but when GW1 fail(I provoke it changing codec) and use failure route (GW 2) then if Client A hang up all works fine, but the problem is when is Client B who hang up, its like a new conversation, GW 2 send BYE to openser and Openser just send "503 Service Not avilable - No gateways" to GW2, but doesnt send nothing to ClientA, any idea ???? Thx in advance this is my basic configuration( I have attached full config file): route{ if (!method=="REGISTER") { record_route(); }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if (method=="INVITE") { if (!lookup("location")) { if (!load_gws()) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); return; }; }; if (!next_gw()) { sl_send_reply("503", "Service not available - No gateways"); return; }; }; }; }; t_on_failure("1"); route(1); } route[1] { if (!t_relay()) { sl_send_reply("666","t_relay fallido"); return; }; exit; } failure_route[1] { if (!next_gw()) { t_reply("503", "Service not available - No gateways 1"); return; }; if (!t_relay()) { t_reply("666","t_relay fallido"); return; }; }

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RE: [Users] Transaction already in Process
by Douglas Garstang 15 Dec '05

15 Dec '05

Klaus, Asterisk can't use a database for location/contact information. It also has no native means to transfer registrations between itself and another Asterisk system. It can't even perform a user defined action when a phone registers... doesn't leave us with many options. Right now, using OpenSER to replicate (forward,t_relay,t_replicate whatever) seems to be working ok. Do you have any reason to believe it wouldn't scale up well? Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Thursday, December 15, 2005 3:23 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Transaction already in Process Hi Douglas! I still think it is a bad idea to replicate REGISTER to Asterisk 4 times. Where does Asterisk store the location table? In a database? Why not replicate the location data on DB level? regards klaus Douglas Garstang wrote: > Arrgh! > > Now I'm getting "t_newtran: transaction already in process" messages whenever I try to call t_relay() or t_replicate twice, even when the first attempt has FAILED. In the case of t_relay() I have new addresses in the destination set and I still get this error. Why? When you attempt to deliver, and it fails, how can to attempt again without getting those messages? What fundamental piece am I missing? > > How could I do this? Forget the fancy stuff. I don't care about checking for failure anymore. > > t_replicate("192.168.10.7","5060"); > t_replicate("192.168.10.8","5060"); > t_replicate("192.168.10.200","5060"); > > Doug. > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

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[Serusers] Weirdness trying to send IM
by sip 15 Dec '05

15 Dec '05

I'm attempting to send an IM via SIP using the method described in the admin guide (from the command line). When I try, I get these messages in the logs: Dec 15 10:10:26 death /usr/local/sbin/ser[8138]: ERROR: fifo_server: command must begin with :: NOTIFY Dec 15 10:10:26 death /usr/local/sbin/ser[8138]: ERROR: fifo_server: command must have at least 3 chars Dec 15 10:10:26 death /usr/local/sbin/ser[8138]: ERROR: fifo_server: command must begin with :: To: sip:1234@my.server.com Dec 15 10:10:26 death /usr/local/sbin/ser[8138]: ERROR: fifo_server: command must begin with :: Hello world!!!! Can someone tell me what crucial step I'm missing here? N.

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RE: [Users] Transaction already in Process
by Douglas Garstang 15 Dec '05

15 Dec '05

Klaus, To answer your earlier question, Asterisk stores the registrations in a local file, using the Berkeley DB format. It's keyed on user/extension. Douglas. -----Original Message----- From: Douglas Garstang Sent: Thursday, December 15, 2005 8:09 AM To: Klaus Darilion Cc: users(a)openser.org Subject: RE: [Users] Transaction already in Process Klaus, Asterisk can't use a database for location/contact information. It also has no native means to transfer registrations between itself and another Asterisk system. It can't even perform a user defined action when a phone registers... doesn't leave us with many options. Right now, using OpenSER to replicate (forward,t_relay,t_replicate whatever) seems to be working ok. Do you have any reason to believe it wouldn't scale up well? Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Thursday, December 15, 2005 3:23 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Transaction already in Process Hi Douglas! I still think it is a bad idea to replicate REGISTER to Asterisk 4 times. Where does Asterisk store the location table? In a database? Why not replicate the location data on DB level? regards klaus Douglas Garstang wrote: > Arrgh! > > Now I'm getting "t_newtran: transaction already in process" messages whenever I try to call t_relay() or t_replicate twice, even when the first attempt has FAILED. In the case of t_relay() I have new addresses in the destination set and I still get this error. Why? When you attempt to deliver, and it fails, how can to attempt again without getting those messages? What fundamental piece am I missing? > > How could I do this? Forget the fancy stuff. I don't care about checking for failure anymore. > > t_replicate("192.168.10.7","5060"); > t_replicate("192.168.10.8","5060"); > t_replicate("192.168.10.200","5060"); > > Doug. > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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RE: [Users] Variables
by Douglas Garstang 15 Dec '05

15 Dec '05

AVP's.... I just can't follow the docs on this. I wish I could. -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Thursday, December 15, 2005 3:43 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Variables Hi Doug, use avp_printf() or avp_subst() to process the content of an avp - this function covers the assigning operation. regards, bogdan Douglas Garstang wrote: >All, > >I can't find any documentation of OpenSER use of variables. I'm sending a lot of info to syslog with xlog(), and with multiple threads (processes?) and multiple calls running, all the output is out of sequence. As a result, I'd like to prefix the debug output with the call-id. The call-id however is very long and I'd like to cut it down and only display a portion of it. > >How can I extract a string from a SIP message, cut it up, extract the portion I want, and print that portion? OpenSER doesn't seem to have any concept of assigning variables by: > >variable = value > >Thanks, >Doug. > > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

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RE: [Users] Variables
by Douglas Garstang 15 Dec '05

15 Dec '05

Oh... those AVP's again which I still completely don't understand.... -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Thursday, December 15, 2005 3:27 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Variables I think xlog can log AVPs. Thus write the callid into an AVP and xlog this AVP. I think you can rewrite the AVP using regexp, thus it should be possible to shorten the callid. klaus Douglas Garstang wrote: > All, > > I can't find any documentation of OpenSER use of variables. I'm sending a lot of info to syslog with xlog(), and with multiple threads (processes?) and multiple calls running, all the output is out of sequence. As a result, I'd like to prefix the debug output with the call-id. The call-id however is very long and I'd like to cut it down and only display a portion of it. > > How can I extract a string from a SIP message, cut it up, extract the portion I want, and print that portion? OpenSER doesn't seem to have any concept of assigning variables by: > > variable = value > > Thanks, > Doug. > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

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Re: [Serusers] using pa module
by lo ™ 15 Dec '05

15 Dec '05

which cvs version should i use? which is the latest stable one? how can i test presence and istant messaging without using eyebeam? ...testing presence often makes eyebeam or ser crash! thank you! On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > > I think that it will work in "peer-to-peer mode" too (I have used > "presence agent mode" in my tests). > And you don't need XCAP or resource lists - your config should be > sufficient. > > It is probably quite old version which had not this fifo command > implemented (and features like resource lists or XCAP auth)... > > You can watch message flows if something working wrong, but I recommend > use of current CVS version (there are still some problems, but it could > work better than old versions). > > Vaclav

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RE: [Users] mysql cluster and openser
by Maria Yndefors 15 Dec '05

15 Dec '05

Ok, I have downloaded the packages from deb-stable directory? So I guess I have to change to the unstable packages... I guess libc is not the same on them. openser_1.0.0.0_i386.deb openser-mysql-module_1.0.0.0_i386.deb Regards /Maria > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: den 15 december 2005 13:07 > To: Maria Yndefors > Cc: users(a)openser.org > Subject: Re: [Users] mysql cluster and openser > > > Hi Maria, > > my question is was for what debian version was compiled the debian > package you are using? I will help me to tell you the > required mysql lib > version. > > regards, > bogdan > > Maria Yndefors wrote: > > >Yes I have testing/unstable on both machines, one which run > SER and one which I > >have installed Openser. > >Regards > >/Maria > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: den 14 december 2005 20:59 > >>To: Maria Yndefors > >>Cc: users(a)openser.org > >>Subject: Re: [Users] mysql cluster and openser > >> > >> > >>Hi Maria, > >> > >>what debian version package are you using? unstable or...? > >> > >>regards, > >>bogdan > >> > >>Maria Yndefors wrote: > >> > >> > >> > >>>Hallo, > >>>I run openser on a debian used the deb package to install it. > >>>I have a ndb-cluster setup that I have used successfully > >>> > >>> > >>with SER 0.9.4, now I have tested to install openser and I > >> > >> > >>>can't manage to get the openser mysql module to find the > >>> > >>> > >>correct libmysql. > >> > >> > >>>I have: > >>> > >>>/usr/lib/libmysqlclient.so.12 > >>>/usr/lib/libmysqlclient.so.15 > >>> > >>>I have tried with a softlink from libmysql.so.10 but I > don't manage. > >>>What is the difference from standard SER and openser mysql module? > >>> > >>>Regards > >>>/Maria > >>> > >>> > >>> > >>>Maria Yndefors > >>>_______________________________________________________________ > >>>System Management & Development > >>>BREDBANDSBOLAGET / B2 Bredband AB > >>>Direct: +46 (0)8 50 69 83 80 > >>>Mobile: +46 (0)7 33 81 92 51 > >>>Fax: +46 (0)8 54 90 46 08 > >>>http://www.bredband.com > >>>_______________________________________________________________ > >>> > >>> > >>>_______________________________________________ > >>>Users mailing list > >>>Users(a)openser.org > >>>http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >>> > >>> > >>> > > > > > > > >

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[Serusers] Help with compiling CVS
by Yong, Chee Yeew 15 Dec '05

15 Dec '05

Hello, First of all, I hope this is the correct forum to post this; if not, please correct me. :) Secondly, I am having problems make-ing the CVS version downloaded from the pserver (sip_router). I have posted the errors below. If anyone can help, please advise. Thanks. Chee ======================== The errors that I am experiencing: /usr/bin/ld: cannot find -lxcap collect2: ld returned 1 exit status make[1]: *** [pa.so] Error 1 avpops.c:138: warning: initialization from incompatible pointer type avpops.c:139: warning: initialization from incompatible pointer type avpops.c:140: warning: initialization from incompatible pointer type avpops.c:141: warning: initialization from incompatible pointer type avpops.c:144: warning: initialization from incompatible pointer type avpops.c: In function `fixup_db_avp': avpops.c:277: error: incompatible types in assignment avpops.c:278: error: invalid operands to binary == avpops.c:282: error: invalid type argument of `->' avpops.c:282: error: cannot convert to a pointer type avpops.c:283: error: invalid type argument of `->' avpops.c:284: error: invalid type argument of `->' avpops.c: In function `fixup_write_avp': avpops.c:429: error: incompatible types in assignment avpops.c:430: error: invalid operands to binary == avpops.c:436: error: invalid type argument of `->' avpops.c:436: error: cannot convert to a pointer type avpops.c:437: error: invalid type argument of `->' avpops.c:438: error: invalid type argument of `->' avpops.c:439: error: invalid type argument of `->' avpops.c:440: error: invalid type argument of `->' avpops.c:440: error: invalid type argument of `->' avpops.c:440: error: invalid type argument of `->' avpops.c:440: error: invalid type argument of `->' avpops.c:440: error: invalid type argument of `->' avpops.c:440: error: invalid type argument of `->' avpops.c:440: error: invalid type argument of `->' avpops.c:440: error: invalid type argument of `->' avpops.c: In function `fixup_pushto_avp': avpops.c:583: error: incompatible types in assignment avpops.c:584: error: invalid operands to binary == avpops.c:589: error: invalid type argument of `->' avpops.c:589: error: cannot convert to a pointer type avpops.c:590: error: invalid type argument of `->' avpops.c:591: error: invalid type argument of `->' avpops.c: In function `fixup_check_avp': avpops.c:679: error: invalid type argument of `->' avpops.c:679: error: invalid type argument of `->' avpops.c:679: error: invalid type argument of `->' avpops.c:679: error: invalid type argument of `->' avpops.c:679: error: invalid type argument of `->' avpops.c:679: error: invalid type argument of `->' avpops.c:679: error: invalid type argument of `->' avpops.c:679: error: invalid type argument of `->' avpops.c:681: error: invalid type argument of `->' avpops.c:684: error: invalid type argument of `->' avpops.c:684: error: invalid type argument of `->' avpops.c:684: error: invalid type argument of `->' avpops.c:684: error: invalid type argument of `->' avpops.c:684: error: invalid type argument of `->' avpops.c:684: error: invalid type argument of `->' avpops.c:684: error: invalid type argument of `->' avpops.c:684: error: invalid type argument of `->' avpops.c:689: error: incompatible type for argument 2 of `qm_free' avpops.c:690: error: incompatible types in assignment make[1]: *** [avpops.o] Error 1 make[1]: *** No targets specified and no makefile found. Stop. make[1]: *** No targets specified and no makefile found. Stop. make[1]: *** No targets specified and no makefile found. Stop. ================================ ******************************************************************************* This email and any files transmitted with it are intended solely for the use of the individual or entity to whom they are addressed and may not be divulged to any third party without the express permission of the originator. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of Thales Research & Technology (UK) Limited. *******************************************************************************

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[Users] AVPops question
by Joachim Fabini 15 Dec '05

15 Dec '05

Hi, Is there a way or workaround to generate AVPs with header values as AVP name? It seems to me that the current AVP concept defines the AVPs at OpenSER startup, so header-value-based names are not feasible. What I need is the following: Store/retrieve a (key,value) pair where the key is the current message's call-id and the value some combination of header values and pseudo- variables. e.g. use the ops avp_printf("$hdr(call-id)", "$hdr(call-id)-$Ts"); avp_pushto("$Myheader","$hdr(call-id)"); in order to append the following header to the SIP message: Myheader: 23459(a)10.0.0.1-4235627623 Any idea how to do this in OpenSER? thanks in advance --Joachim

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RE: [Users] mysql cluster and openser
by Maria Yndefors 15 Dec '05

15 Dec '05

Yes I have testing/unstable on both machines, one which run SER and one which I have installed Openser. Regards /Maria > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: den 14 december 2005 20:59 > To: Maria Yndefors > Cc: users(a)openser.org > Subject: Re: [Users] mysql cluster and openser > > > Hi Maria, > > what debian version package are you using? unstable or...? > > regards, > bogdan > > Maria Yndefors wrote: > > >Hallo, > >I run openser on a debian used the deb package to install it. > >I have a ndb-cluster setup that I have used successfully > with SER 0.9.4, now I have tested to install openser and I > >can't manage to get the openser mysql module to find the > correct libmysql. > > > >I have: > > > >/usr/lib/libmysqlclient.so.12 > >/usr/lib/libmysqlclient.so.15 > > > >I have tried with a softlink from libmysql.so.10 but I don't manage. > >What is the difference from standard SER and openser mysql module? > > > >Regards > >/Maria > > > > > > > >Maria Yndefors > >_______________________________________________________________ > >System Management & Development > >BREDBANDSBOLAGET / B2 Bredband AB > >Direct: +46 (0)8 50 69 83 80 > >Mobile: +46 (0)7 33 81 92 51 > >Fax: +46 (0)8 54 90 46 08 > >http://www.bredband.com > >_______________________________________________________________ > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

2 1

Re: [Serusers] need help with ser-0.9.4 and SUBSCRIBE requests
by Vaclav Kubart 15 Dec '05

15 Dec '05

Hi, I recommend the CVS version or - may be better - latest version from "presence snapshot" (ftp://ftp.iptel.org/pub/ser/presence/). Vaclav On Thu, 15 Dec 2005, Ekaterina Ivannikova wrote: > Hello, > > I just tried Windows Messenger 5.1 (tcpdump follows), but ser produces the > same error as for 4.7. Could the list members please advise which version > of ser should I use to make SUBSCRIBE requests of WM work ? > > Regards, > > Ekaterina > > ~~~~~~~~~~~~~~~~~~ > REGISTER sip:gutatelecom.ru SIP/2.0 > Via: SIP/2.0/UDP 10.123.0.190:12479 > Max-Forwards: 70 > From: <sip:kate@gutatelecom.ru>;tag=3f7630c4da1b459ea8785c10c6c29676;epid=6ffbeddc9d > To: <sip:kate@gutatelecom.ru> > Call-ID: 9d356b245a924154bb96a2a43144add1 > CSeq: 1 REGISTER > Contact: <sip:10.123.0.190:12479>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER, BENOTIFY" > User-Agent: RTC/1.3.5470 (Messenger 5.1.0701) > Event: registration > Allow-Events: presence > Content-Length: 0 > C6 > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.123.0.190:12479 > From: <sip:kate@gutatelecom.ru>;tag=3f7630c4da1b459ea8785c10c6c29676;epid=6ffbeddc9d > To: <sip:kate@gutatelecom.ru>;tag=9bf9a4f66ee3c16d13ef4422214649c2.f220 > Call-ID: 9d356b245a924154bb96a2a43144add1 > CSeq: 1 REGISTER > Contact: <sip:10.123.0.190:7808>;expires=3288, <sip:10.123.0.190:12479>;expires=3600 > Server: Sip EXpress router (0.9.4 (i386/solaris)) > Content-Length: 0 > Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=25299 req_src_ip=10.123.0.190 req_src_port=1041 in_uri=sip:gutatelecom.ru out_uri=sip:gutatelecom.ru via_cnt==1" > C$# > 2 ?SUBSCRIBE sip:ura@gutatelecom.ru SIP/2.0 > Via: SIP/2.0/UDP 10.123.0.190:12479 > Max-Forwards: 70 > From: "kate(a)gutatelecom.ru" <sip:kate@gutatelecom.ru>;tag=1fe50fffda3b490192c4a4e5badd2008;epid=6ffbeddc9d > To: <sip:ura@gutatelecom.ru> > Call-ID: 4ab889c390a14810aea197bb705e2e9d > CSeq: 1 SUBSCRIBE > Contact: <sip:10.123.0.190:12479> > User-Agent: RTC/1.3 > Event: presence > Accept: application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml > Supported: com.microsoft.autoextend > Supported: ms-benotify > Proxy-Require: ms-benotify > Content-Length: 0 > C~# > 2":SUBSCRIBE sip:admin@gutagroup.ru SIP/2.0 > Via: SIP/2.0/UDP 10.123.0.190:12479 > Max-Forwards: 70 > From: "kate(a)gutatelecom.ru" <sip:kate@gutatelecom.ru>;tag=40649ffd5fb04089a23bdb25fba6ed3f;epid=6ffbeddc9d > To: <sip:admin@gutagroup.ru> > Call-ID: 9f52e3dd37ad46e79b0517f06e878b67 > CSeq: 1 SUBSCRIBE > Contact: <sip:10.123.0.190:12479> > User-Agent: RTC/1.3 > Event: presence > Accept: application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml > Supported: com.microsoft.autoextend > Supported: ms-benotify > Proxy-Require: ms-benotify > Content-Length: 0 > vSIP/2.0 400 Bad Request > Via: SIP/2.0/UDP 10.123.0.190:12479 > From: "kate(a)gutatelecom.ru" <sip:kate@gutatelecom.ru>;tag=1fe50fffda3b490192c4a4e5badd2008;epid=6ffbeddc9d > To: <sip:ura@gutatelecom.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-f220 > Call-ID: 4ab889c390a14810aea197bb705e2e9d > CSeq: 1 SUBSCRIBE > Error while parsing headersServer: Sip EXpress router (0.9.4 (i386/solaris)) > Content-Length: 0 > Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=25299 req_src_ip=10.123.0.190 req_src_port=1041 in_uri=sip:ura@gutatelecom.ru out_uri=sip:ura@gutatelecom.ru via_cnt==1" > C~+ > oSIP/2.0 400 Bad Request > Via: SIP/2.0/UDP 10.123.0.190:12479 > From: "kate(a)gutatelecom.ru" <sip:kate@gutatelecom.ru>;tag=40649ffd5fb04089a23bdb25fba6ed3f;epid=6ffbeddc9d > To: <sip:admin@gutagroup.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-f220 > Call-ID: 9f52e3dd37ad46e79b0517f06e878b67 > CSeq: 1 SUBSCRIBE > Error while parsing headersServer: Sip EXpress router (0.9.4 (i386/solaris)) > Content-Length: 0 > Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=25299 req_src_ip=10.123.0.190 req_src_port=1041 in_uri=sip:admin@gutagroup.ru out_uri=sip:admin@gutagroup.ru via_cnt==1" > > On Thu, Dec 15, 2005 at 09:33:18AM +0100, Vaclav Kubart wrote: >> Hello, >> I'm quite sure, that SER was able to work with Windows Messenger 4.7 (it >> is able to work with version 5.1), but I'm not sure about the version >> 0.9.4. May be that it wasn't in that version. >> >> Vaclav >> >> On Thu, 15 Dec 2005, Ekaterina Ivannikova wrote: >> >>> Hello list, >>> >>> I installed a binary package of ser-0.9.4 on Solaris 10 (Intel) from >>> ftp.berlios.de and am trying to use Windows Messenger 4.7 as a client. >>> However SUBSCRIBE requests fail with an obscure message "Error while >>> parsing >>> headers". Googling found several similar complaints, but I still can't >>> figure >>> out how to resolve the issue. Any help is very much appreciated. My config, >>> syslog messages and tcpdump follow. >>> >>> Regards, >>> >>> Ekaterina >>> >>> # ----------- global configuration parameters ------------------------ >>> >>> #debug=3 # debug level (cmd line: -dddddddddd) >>> #fork=yes >>> #log_stderror=no # (cmd line: -E) >>> >>> /* Uncomment these lines to enter debugging mode >>> fork=no >>> log_stderror=yes >>> */ >>> >>> check_via=no # (cmd. line: -v) >>> dns=no # (cmd. line: -r) >>> rev_dns=no # (cmd. line: -R) >>> #port=5060 >>> #children=4 >>> >>> listen=10.123.0.250 >>> user=ser >>> group=ser >>> >>> fifo="/tmp/ser_fifo" >>> >>> # ------------------ module loading ---------------------------------- >>> >>> # Uncomment this if you want to use SQL database >>> loadmodule "/usr/local/lib/ser/modules/mysql.so" >>> loadmodule "/usr/local/lib/ser/modules/sl.so" >>> loadmodule "/usr/local/lib/ser/modules/tm.so" >>> loadmodule "/usr/local/lib/ser/modules/rr.so" >>> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >>> loadmodule "/usr/local/lib/ser/modules/usrloc.so" >>> loadmodule "/usr/local/lib/ser/modules/registrar.so" >>> loadmodule "/usr/local/lib/ser/modules/textops.so" >>> loadmodule "/usr/local/lib/ser/modules/pa.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" >>> >>> # ----------------- setting module-specific parameters --------------- >>> >>> # -- usrloc params -- >>> >>> #modparam("usrloc", "db_mode", 0) >>> >>> # Uncomment this if you want to use SQL database >>> # for persistent storage and comment the previous line >>> modparam("usrloc", "db_mode", 2) >>> >>> # -- auth params -- >>> # Uncomment if you are using auth module >>> # >>> #modparam("auth_db", "calculate_ha1", yes) >>> # >>> # If you set "calculate_ha1" parameter to yes (which true in this config), >>> # uncomment also the following parameter) >>> # >>> #modparam("auth_db", "password_column", "password") >>> >>> # -- rr params -- >>> # add value to ;lr param to make some broken UAs happy >>> modparam("rr", "enable_full_lr", 1) >>> >>> modparam("usrloc|registrar", "use_domain", 1) >>> modparam("pa", "use_db", 1) >>> modparam("pa", "db_url", "mysql://ser:heslo@localhost/ser") >>> modparam("pa", "pa_domain", "gutatelecom.ru") >>> >>> >>> # ------------------------- request routing logic ------------------- >>> >>> # main routing logic >>> >>> alias="z1.all.yans.ru" >>> alias="gutatelecom.ru" >>> alias="gutagroup.ru" >>> alias="10.123.0.250" >>> >>> 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("iptel.org", "subscriber")) { >>> # www_challenge("iptel.org", "0"); >>> # break; >>> # }; >>> >>> save("location"); >>> 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(); >>> }; >>> } >>> >>> >>> Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] >>> setting signal 2 to 8063eb0 >>> Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] >>> setting signal 13 to 8063eb0 >>> Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] >>> setting signal 16 to 8063eb0 >>> Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] >>> setting signal 18 to 8063eb0 >>> Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] >>> setting signal 15 to 8063eb0 >>> Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] >>> setting signal 1 to 8063eb0 >>> Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] >>> setting signal 17 to 8063eb0 >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 212014 daemon.warning] >>> Warning: uac does not spread across the whole hash table >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 499528 daemon.notice] >>> Maxfwd module- initializing >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 520766 daemon.crit] >>> db_url=8228c90 >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 520769 daemon.crit] >>> db_url=mysql://ser:heslo@localhost/ser >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 472096 daemon.crit] >>> db_url.len=31 >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 351855 daemon.crit] >>> pa_domain=gutatelecom.ru >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 624776 daemon.crit] >>> pa_mod: use_db=1 db_url.s=mysql://ser:heslo@localhost/ser >>> pa_domain=gutatelecom.ru >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 101486 daemon.crit] >>> pa_mod_init done >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 187949 daemon.error] >>> subscribe_fixup: pdomain name is registrar >>> Dec 13 17:16:15 z1 last message repeated 1 time >>> Dec 13 17:16:15 z1 /usr/local/sbin/ser[24587]: [ID 827449 daemon.warning] >>> WARNING: no fifo_db_url given - fifo DB commands disabled! >>> Dec 13 17:17:25 z1 /usr/local/sbin/ser[24588]: [ID 219598 daemon.error] >>> handle_subscription() entered >>> Dec 13 17:17:25 z1 /usr/local/sbin/ser[24588]: [ID 717776 daemon.error] >>> parse_hfs(): Error while parsing headers >>> Dec 13 17:17:25 z1 /usr/local/sbin/ser[24588]: [ID 828236 daemon.error] >>> handle_subscription(): Error while parsing message header >>> Dec 13 17:17:25 z1 /usr/local/sbin/ser[24588]: [ID 405054 daemon.error] >>> handle_subscription about to send_reply and return -2 >>> Dec 13 17:17:25 z1 /usr/local/sbin/ser[24590]: [ID 219598 daemon.error] >>> handle_subscription() entered >>> Dec 13 17:17:25 z1 /usr/local/sbin/ser[24590]: [ID 717776 daemon.error] >>> parse_hfs(): Error while parsing headers >>> Dec 13 17:17:25 z1 /usr/local/sbin/ser[24590]: [ID 828236 daemon.error] >>> handle_subscription(): Error while parsing message header >>> Dec 13 17:17:25 z1 /usr/local/sbin/ser[24590]: [ID 405054 daemon.error] >>> handle_subscription about to send_reply and return -2 >>> Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 219598 daemon.error] >>> handle_subscription() entered >>> Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 717776 daemon.error] >>> parse_hfs(): Error while parsing headers >>> Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 828236 daemon.error] >>> handle_subscription(): Error while parsing message header >>> Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 405054 daemon.error] >>> handle_subscription about to send_reply and return -2 >>> Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 219598 daemon.error] >>> handle_subscription() entered >>> Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 717776 daemon.error] >>> parse_hfs(): Error while parsing headers >>> Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 828236 daemon.error] >>> handle_subscription(): Error while parsing message header >>> Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 405054 daemon.error] >>> handle_subscription about to send_reply and return -2 >>> >>> ~~~~~~~~~~~~~~~~~~ >>> REGISTER sip:gutatelecom.ru SIP/2.0 >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> Max-Forwards: 70 >>> From: >>> <sip:kate@gutatelecom.ru>;tag=f3dda05188a9449abba6930f6dc9ba57;epid=d631b39a7b >>> To: <sip:kate@gutatelecom.ru> >>> Call-ID: fc49c5f7ddbd4bf78718b0b11986caf4(a)10.123.0.190 >>> CSeq: 1 REGISTER >>> Contact: <sip:10.123.0.190:10707>;methods="INVITE, MESSAGE, INFO, >>> SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" >>> User-Agent: RTC/1.2.4949 >>> Event: registration >>> Allow-Events: presence >>> Content-Length: 0 >>> K)^SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> From: >>> <sip:kate@gutatelecom.ru>;tag=f3dda05188a9449abba6930f6dc9ba57;epid=d631b39a7b >>> To: <sip:kate@gutatelecom.ru>;tag=9bf9a4f66ee3c16d13ef4422214649c2.b407 >>> Call-ID: fc49c5f7ddbd4bf78718b0b11986caf4(a)10.123.0.190 >>> CSeq: 1 REGISTER >>> Contact: <sip:10.123.0.190:10707>;expires=3600 >>> Server: Sip EXpress router (0.9.4 (i386/solaris)) >>> Content-Length: 0 >>> Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24588 >>> req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:gutatelecom.ru >>> out_uri=sip:gutatelecom.ru via_cnt==1" >>> SUBSCRIBE sip:ura@gutatelecom.ru SIP/2.0 >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> Max-Forwards: 70 >>> From: "kate" >>> <sip:kate@gutatelecom.ru>;tag=88b65cf1039140ba8f3fb3a0177ad1f0;epid=d631b39a7b >>> To: <sip:ura@gutatelecom.ru> >>> Call-ID: bf50c9641de74282a1f950e114e446de(a)10.123.0.190 >>> CSeq: 1 SUBSCRIBE >>> Contact: <sip:10.123.0.190:10707> >>> User-Agent: RTC/1.2 >>> Event: presence >>> Accept: application/xpidf+xml, text/xml+msrtc.pidf >>> Supported: com.microsoft.autoextend >>> Content-Length: 0 >>> SUBSCRIBE sip:admin@gutagroup.ru SIP/2.0 >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> Max-Forwards: 70 >>> From: "kate" >>> <sip:kate@gutatelecom.ru>;tag=28b82b1b69e34ffda395eafa53bff1fa;epid=d631b39a7b >>> To: <sip:admin@gutagroup.ru> >>> Call-ID: 0c95f18aac8a45578e90b1f61e5ad634(a)10.123.0.190 >>> CSeq: 1 SUBSCRIBE >>> Contact: <sip:10.123.0.190:10707> >>> User-Agent: RTC/1.2 >>> Event: presence >>> Accept: application/xpidf+xml, text/xml+msrtc.pidf >>> Supported: com.microsoft.autoextend >>> Content-Length: 0 >>> SIP/2.0 400 Bad Request >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> From: "kate" >>> <sip:kate@gutatelecom.ru>;tag=28b82b1b69e34ffda395eafa53bff1fa;epid=d631b39a7b >>> To: <sip:admin@gutagroup.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-b407 >>> Call-ID: 0c95f18aac8a45578e90b1f61e5ad634(a)10.123.0.190 >>> CSeq: 1 SUBSCRIBE >>> Error while parsing headersServer: Sip EXpress router (0.9.4 >>> (i386/solaris)) >>> Content-Length: 0 >>> Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24588 >>> req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:admin@gutagroup.ru >>> out_uri=sip:admin@gutagroup.ru via_cnt==1" >>> SIP/2.0 400 Bad Request >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> From: "kate" >>> <sip:kate@gutatelecom.ru>;tag=88b65cf1039140ba8f3fb3a0177ad1f0;epid=d631b39a7b >>> To: <sip:ura@gutatelecom.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-b407 >>> Call-ID: bf50c9641de74282a1f950e114e446de(a)10.123.0.190 >>> CSeq: 1 SUBSCRIBE >>> Error while parsing headersServer: Sip EXpress router (0.9.4 >>> (i386/solaris)) >>> Content-Length: 0 >>> Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24590 >>> req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:ura@gutatelecom.ru >>> out_uri=sip:ura@gutatelecom.ru via_cnt==1" >>> SUBSCRIBE sip:ura@gutatelecom.ru SIP/2.0 >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> Max-Forwards: 70 >>> From: "kate" >>> <sip:kate@gutatelecom.ru>;tag=88b65cf1039140ba8f3fb3a0177ad1f0;epid=d631b39a7b >>> To: <sip:ura@gutatelecom.ru> >>> Call-ID: bf50c9641de74282a1f950e114e446de(a)10.123.0.190 >>> CSeq: 1 SUBSCRIBE >>> Contact: <sip:10.123.0.190:10707> >>> User-Agent: RTC/1.2 >>> Event: presence >>> Accept: application/xpidf+xml, text/xml+msrtc.pidf >>> Supported: com.microsoft.autoextend >>> Content-Length: 0 >>> SUBSCRIBE sip:admin@gutagroup.ru SIP/2.0 >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> Max-Forwards: 70 >>> From: "kate" >>> <sip:kate@gutatelecom.ru>;tag=28b82b1b69e34ffda395eafa53bff1fa;epid=d631b39a7b >>> To: <sip:admin@gutagroup.ru> >>> Call-ID: 0c95f18aac8a45578e90b1f61e5ad634(a)10.123.0.190 >>> CSeq: 1 SUBSCRIBE >>> Contact: <sip:10.123.0.190:10707> >>> User-Agent: RTC/1.2 >>> Event: presence >>> Accept: application/xpidf+xml, text/xml+msrtc.pidf >>> Supported: com.microsoft.autoextend >>> Content-Length: 0 >>> SIP/2.0 400 Bad Request >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> From: "kate" >>> <sip:kate@gutatelecom.ru>;tag=88b65cf1039140ba8f3fb3a0177ad1f0;epid=d631b39a7b >>> To: <sip:ura@gutatelecom.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-b407 >>> Call-ID: bf50c9641de74282a1f950e114e446de(a)10.123.0.190 >>> CSeq: 1 SUBSCRIBE >>> Error while parsing headersServer: Sip EXpress router (0.9.4 >>> (i386/solaris)) >>> Content-Length: 0 >>> Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24590 >>> req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:ura@gutatelecom.ru >>> out_uri=sip:ura@gutatelecom.ru via_cnt==1" >>> SIP/2.0 400 Bad Request >>> Via: SIP/2.0/UDP 10.123.0.190:10707 >>> From: "kate" >>> <sip:kate@gutatelecom.ru>;tag=28b82b1b69e34ffda395eafa53bff1fa;epid=d631b39a7b >>> To: <sip:admin@gutagroup.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-b407 >>> Call-ID: 0c95f18aac8a45578e90b1f61e5ad634(a)10.123.0.190 >>> CSeq: 1 SUBSCRIBE >>> Error while parsing headersServer: Sip EXpress router (0.9.4 >>> (i386/solaris)) >>> Content-Length: 0 >>> Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24588 >>> req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:admin@gutagroup.ru >>> out_uri=sip:admin@gutagroup.ru via_cnt==1" >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >

1 0

[Serusers] need help with ser-0.9.4 and SUBSCRIBE requests
by Ekaterina Ivannikova 15 Dec '05

15 Dec '05

Hello list, I installed a binary package of ser-0.9.4 on Solaris 10 (Intel) from ftp.berlios.de and am trying to use Windows Messenger 4.7 as a client. However SUBSCRIBE requests fail with an obscure message "Error while parsing headers". Googling found several similar complaints, but I still can't figure out how to resolve the issue. Any help is very much appreciated. My config, syslog messages and tcpdump follow. Regards, Ekaterina # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 listen=10.123.0.250 user=ser group=ser fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/pa.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" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("usrloc|registrar", "use_domain", 1) modparam("pa", "use_db", 1) modparam("pa", "db_url", "mysql://ser:heslo@localhost/ser") modparam("pa", "pa_domain", "gutatelecom.ru") # ------------------------- request routing logic ------------------- # main routing logic alias="z1.all.yans.ru" alias="gutatelecom.ru" alias="gutagroup.ru" alias="10.123.0.250" 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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); 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(); }; } Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] setting signal 2 to 8063eb0 Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] setting signal 13 to 8063eb0 Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] setting signal 16 to 8063eb0 Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] setting signal 18 to 8063eb0 Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] setting signal 15 to 8063eb0 Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] setting signal 1 to 8063eb0 Dec 13 17:16:14 z1 /usr/local/sbin/ser[24585]: [ID 326729 daemon.crit] setting signal 17 to 8063eb0 Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 212014 daemon.warning] Warning: uac does not spread across the whole hash table Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 499528 daemon.notice] Maxfwd module- initializing Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 520766 daemon.crit] db_url=8228c90 Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 520769 daemon.crit] db_url=mysql://ser:heslo@localhost/ser Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 472096 daemon.crit] db_url.len=31 Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 351855 daemon.crit] pa_domain=gutatelecom.ru Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 624776 daemon.crit] pa_mod: use_db=1 db_url.s=mysql://ser:heslo@localhost/ser pa_domain=gutatelecom.ru Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 101486 daemon.crit] pa_mod_init done Dec 13 17:16:15 z1 /usr/local/sbin/ser[24585]: [ID 187949 daemon.error] subscribe_fixup: pdomain name is registrar Dec 13 17:16:15 z1 last message repeated 1 time Dec 13 17:16:15 z1 /usr/local/sbin/ser[24587]: [ID 827449 daemon.warning] WARNING: no fifo_db_url given - fifo DB commands disabled! Dec 13 17:17:25 z1 /usr/local/sbin/ser[24588]: [ID 219598 daemon.error] handle_subscription() entered Dec 13 17:17:25 z1 /usr/local/sbin/ser[24588]: [ID 717776 daemon.error] parse_hfs(): Error while parsing headers Dec 13 17:17:25 z1 /usr/local/sbin/ser[24588]: [ID 828236 daemon.error] handle_subscription(): Error while parsing message header Dec 13 17:17:25 z1 /usr/local/sbin/ser[24588]: [ID 405054 daemon.error] handle_subscription about to send_reply and return -2 Dec 13 17:17:25 z1 /usr/local/sbin/ser[24590]: [ID 219598 daemon.error] handle_subscription() entered Dec 13 17:17:25 z1 /usr/local/sbin/ser[24590]: [ID 717776 daemon.error] parse_hfs(): Error while parsing headers Dec 13 17:17:25 z1 /usr/local/sbin/ser[24590]: [ID 828236 daemon.error] handle_subscription(): Error while parsing message header Dec 13 17:17:25 z1 /usr/local/sbin/ser[24590]: [ID 405054 daemon.error] handle_subscription about to send_reply and return -2 Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 219598 daemon.error] handle_subscription() entered Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 717776 daemon.error] parse_hfs(): Error while parsing headers Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 828236 daemon.error] handle_subscription(): Error while parsing message header Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 405054 daemon.error] handle_subscription about to send_reply and return -2 Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 219598 daemon.error] handle_subscription() entered Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 717776 daemon.error] parse_hfs(): Error while parsing headers Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 828236 daemon.error] handle_subscription(): Error while parsing message header Dec 13 17:17:32 z1 /usr/local/sbin/ser[24593]: [ID 405054 daemon.error] handle_subscription about to send_reply and return -2 ~~~~~~~~~~~~~~~~~~ REGISTER sip:gutatelecom.ru SIP/2.0 Via: SIP/2.0/UDP 10.123.0.190:10707 Max-Forwards: 70 From: <sip:kate@gutatelecom.ru>;tag=f3dda05188a9449abba6930f6dc9ba57;epid=d631b39a7b To: <sip:kate@gutatelecom.ru> Call-ID: fc49c5f7ddbd4bf78718b0b11986caf4(a)10.123.0.190 CSeq: 1 REGISTER Contact: <sip:10.123.0.190:10707>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" User-Agent: RTC/1.2.4949 Event: registration Allow-Events: presence Content-Length: 0 K)^SIP/2.0 200 OK Via: SIP/2.0/UDP 10.123.0.190:10707 From: <sip:kate@gutatelecom.ru>;tag=f3dda05188a9449abba6930f6dc9ba57;epid=d631b39a7b To: <sip:kate@gutatelecom.ru>;tag=9bf9a4f66ee3c16d13ef4422214649c2.b407 Call-ID: fc49c5f7ddbd4bf78718b0b11986caf4(a)10.123.0.190 CSeq: 1 REGISTER Contact: <sip:10.123.0.190:10707>;expires=3600 Server: Sip EXpress router (0.9.4 (i386/solaris)) Content-Length: 0 Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24588 req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:gutatelecom.ru out_uri=sip:gutatelecom.ru via_cnt==1" SUBSCRIBE sip:ura@gutatelecom.ru SIP/2.0 Via: SIP/2.0/UDP 10.123.0.190:10707 Max-Forwards: 70 From: "kate" <sip:kate@gutatelecom.ru>;tag=88b65cf1039140ba8f3fb3a0177ad1f0;epid=d631b39a7b To: <sip:ura@gutatelecom.ru> Call-ID: bf50c9641de74282a1f950e114e446de(a)10.123.0.190 CSeq: 1 SUBSCRIBE Contact: <sip:10.123.0.190:10707> User-Agent: RTC/1.2 Event: presence Accept: application/xpidf+xml, text/xml+msrtc.pidf Supported: com.microsoft.autoextend Content-Length: 0 SUBSCRIBE sip:admin@gutagroup.ru SIP/2.0 Via: SIP/2.0/UDP 10.123.0.190:10707 Max-Forwards: 70 From: "kate" <sip:kate@gutatelecom.ru>;tag=28b82b1b69e34ffda395eafa53bff1fa;epid=d631b39a7b To: <sip:admin@gutagroup.ru> Call-ID: 0c95f18aac8a45578e90b1f61e5ad634(a)10.123.0.190 CSeq: 1 SUBSCRIBE Contact: <sip:10.123.0.190:10707> User-Agent: RTC/1.2 Event: presence Accept: application/xpidf+xml, text/xml+msrtc.pidf Supported: com.microsoft.autoextend Content-Length: 0 SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 10.123.0.190:10707 From: "kate" <sip:kate@gutatelecom.ru>;tag=28b82b1b69e34ffda395eafa53bff1fa;epid=d631b39a7b To: <sip:admin@gutagroup.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-b407 Call-ID: 0c95f18aac8a45578e90b1f61e5ad634(a)10.123.0.190 CSeq: 1 SUBSCRIBE Error while parsing headersServer: Sip EXpress router (0.9.4 (i386/solaris)) Content-Length: 0 Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24588 req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:admin@gutagroup.ru out_uri=sip:admin@gutagroup.ru via_cnt==1" SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 10.123.0.190:10707 From: "kate" <sip:kate@gutatelecom.ru>;tag=88b65cf1039140ba8f3fb3a0177ad1f0;epid=d631b39a7b To: <sip:ura@gutatelecom.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-b407 Call-ID: bf50c9641de74282a1f950e114e446de(a)10.123.0.190 CSeq: 1 SUBSCRIBE Error while parsing headersServer: Sip EXpress router (0.9.4 (i386/solaris)) Content-Length: 0 Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24590 req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:ura@gutatelecom.ru out_uri=sip:ura@gutatelecom.ru via_cnt==1" SUBSCRIBE sip:ura@gutatelecom.ru SIP/2.0 Via: SIP/2.0/UDP 10.123.0.190:10707 Max-Forwards: 70 From: "kate" <sip:kate@gutatelecom.ru>;tag=88b65cf1039140ba8f3fb3a0177ad1f0;epid=d631b39a7b To: <sip:ura@gutatelecom.ru> Call-ID: bf50c9641de74282a1f950e114e446de(a)10.123.0.190 CSeq: 1 SUBSCRIBE Contact: <sip:10.123.0.190:10707> User-Agent: RTC/1.2 Event: presence Accept: application/xpidf+xml, text/xml+msrtc.pidf Supported: com.microsoft.autoextend Content-Length: 0 SUBSCRIBE sip:admin@gutagroup.ru SIP/2.0 Via: SIP/2.0/UDP 10.123.0.190:10707 Max-Forwards: 70 From: "kate" <sip:kate@gutatelecom.ru>;tag=28b82b1b69e34ffda395eafa53bff1fa;epid=d631b39a7b To: <sip:admin@gutagroup.ru> Call-ID: 0c95f18aac8a45578e90b1f61e5ad634(a)10.123.0.190 CSeq: 1 SUBSCRIBE Contact: <sip:10.123.0.190:10707> User-Agent: RTC/1.2 Event: presence Accept: application/xpidf+xml, text/xml+msrtc.pidf Supported: com.microsoft.autoextend Content-Length: 0 SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 10.123.0.190:10707 From: "kate" <sip:kate@gutatelecom.ru>;tag=88b65cf1039140ba8f3fb3a0177ad1f0;epid=d631b39a7b To: <sip:ura@gutatelecom.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-b407 Call-ID: bf50c9641de74282a1f950e114e446de(a)10.123.0.190 CSeq: 1 SUBSCRIBE Error while parsing headersServer: Sip EXpress router (0.9.4 (i386/solaris)) Content-Length: 0 Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24590 req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:ura@gutatelecom.ru out_uri=sip:ura@gutatelecom.ru via_cnt==1" SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 10.123.0.190:10707 From: "kate" <sip:kate@gutatelecom.ru>;tag=28b82b1b69e34ffda395eafa53bff1fa;epid=d631b39a7b To: <sip:admin@gutagroup.ru>;tag=4847b256b8606eb0a98fbd0f755585cd-b407 Call-ID: 0c95f18aac8a45578e90b1f61e5ad634(a)10.123.0.190 CSeq: 1 SUBSCRIBE Error while parsing headersServer: Sip EXpress router (0.9.4 (i386/solaris)) Content-Length: 0 Warning: 392 10.123.0.250:5060 "Noisy feedback tells: pid=24588 req_src_ip=10.123.0.190 req_src_port=1271 in_uri=sip:admin@gutagroup.ru out_uri=sip:admin@gutagroup.ru via_cnt==1"

2 2

[Users] login problem
by unplug 15 Dec '05

15 Dec '05

I am using mysql.so for the user login. User can login without problem. After a few hours, user issues still issue a register request as below but can't login to the system (there is no record in the db). Why user failed to login suddently? I am sure the username and password are correct. From the log, I can see the register request as below. However, user login failed. Dec 14 17:19:32 os01 /usr/local/sbin/openser[14889]: SIP Request: method [REGISTER] from [210.202.7.17] [1833] to [203.193.146.6] [5060]

2 4

[Serusers] SER-0.9.4 and Mysql 5.0.15
by han yibing 15 Dec '05

15 Dec '05

Hello All: I am using SER-0.9.4 with mysql 5.0.15,when i create database and tables with following error: [root@localhost sbin]# ./ser_mysql.sh create MySql password for root: creating database ser ... ERROR 1049 (42000) at line 2: Unknown database 'ser;' please help me !!! thanks __________________________________________________ 赶快注册雅虎超大容量免费邮箱? http://cn.mail.yahoo.com

3 2

[Serusers] problems with ser + radius
by Guido 15 Dec '05

15 Dec '05

Hello, I'm using fedora core 4 and radiusclient-0.4.3. installing SER 0.9.4 with support for radius I can't install radius modules and when run my SER I view this mesasge error 0(441) ERROR: load_module: could not open module </usr/local/lib/ser/modules/auth_radius.so>: /usr/local/lib/ser/modules/auth_radius.so: cannot open shared object file: No such file or directory 0(441) parse error (41,13-14): failed to load module 0(441) set_mod_param_regex: No module matching auth_radius found | 0(441) parse error (71,20-21): Can't set module parameter 0(441) set_mod_param_regex: No module matching auth_radius found | 0(441) parse error (72,20-21): Can't set module parameter 0(441) parse error (326,34-35): unknown command, missing loadmodule? Anybody can help me ? Thanks Guido

2 1

[Users] Variables
by Douglas Garstang 15 Dec '05

15 Dec '05

All, I can't find any documentation of OpenSER use of variables. I'm sending a lot of info to syslog with xlog(), and with multiple threads (processes?) and multiple calls running, all the output is out of sequence. As a result, I'd like to prefix the debug output with the call-id. The call-id however is very long and I'd like to cut it down and only display a portion of it. How can I extract a string from a SIP message, cut it up, extract the portion I want, and print that portion? OpenSER doesn't seem to have any concept of assigning variables by: variable = value Thanks, Doug.

3 2

[Users] How to send SIP REFER?
by James H Thompson 15 Dec '05

15 Dec '05

Whats the best way to send out a SIP REFER in reply to a received INVITE? Thanks.

4 4

[Users] Problem in creating MySQL database
by Neil 15 Dec '05

15 Dec '05

Hi all, I am having problem in creating MySQL database, I followed the given below instruction create MySQL tables - if you have a previously installed OpenSER on your system, use /usr/sbin/openser_mysql.sh reinstall to convert your OpenSER database into new structures - otherwise, if this is your very first installation, use /usr/sbin/openser_mysql.sh create to create OpenSER database structures (you will be prompted for password of MySql "root" user) but when i tried to create it, this is what happened /sbin/openser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': admin /sbin/openser_mysql.sh: line 201: openser_gen_ha1: command not found HA1 calculation failed any help regarding this would be favourable thanks naveen

4 5

[Users] Transaction already in Process
by Douglas Garstang 15 Dec '05

15 Dec '05

Arrgh! Now I'm getting "t_newtran: transaction already in process" messages whenever I try to call t_relay() or t_replicate twice, even when the first attempt has FAILED. In the case of t_relay() I have new addresses in the destination set and I still get this error. Why? When you attempt to deliver, and it fails, how can to attempt again without getting those messages? What fundamental piece am I missing? How could I do this? Forget the fancy stuff. I don't care about checking for failure anymore. t_replicate("192.168.10.7","5060"); t_replicate("192.168.10.8","5060"); t_replicate("192.168.10.200","5060"); Doug.

3 2

[Serusers] FreeBSD 5.3 replace ["$2" == "" ] wirh ["$2" = "" ] in serctl.
by Richard Lamb 15 Dec '05

15 Dec '05

In response to below. replace ["$2" == "" ] wirh ["$2" = "" ] in serctl. Too much "C" think.. Richard Message: 1 Date: Wed, 02 Mar 2005 14:03:08 +0800 From: " " Subject: [Serusers] FREEBSD ser cvs adn 0.90 ERROR To: serusers at iptel.org Message-ID: <20050302060308.CCE8F13D at mail.fivewall.com> Content-Type: text/plain; charset= Hi! all FreeBSD 5.3,latest cvs and 0.90 have problem when /usr/local/sbin/serctl get ser# ps -ax PID TT STAT TIME COMMAND 0 ?? DLs 0:00.01 [swapper] ...clip... 543 v7 Is+ 0:00.01 /usr/libexec/getty Pc ttyv7 443 con- I 0:00.04 /bin/sh /usr/local/bin/mysqld_safe --user=mysql --dat 468 con- S 0:01.00 /usr/local/libexec/mysqld --basedir=/usr/local --data ser# serctl moni [: unexpected operator [: -ne: unexpected operator ser# ---- ZhongShan Ether Network Security Inc

2 1

Re: [Serusers] SIP call hold/unhold
by samuel 15 Dec '05

15 Dec '05

Sending to the list as more people might be interested... I read offer/answer RFC quite a long ago but as far as I remember there is no limit in the parameters you can change in a new offer/answer exchange. According to this, UA1 can change the port and the media direccion in the same offer/answer exchange. Samuel. 2005/12/15, tcchan <tikchoong.chan(a)redtone.com>: > Dear Samuel, > > > Thanks you very much for your answer. > > > I would like to consult you the following: > > normal call, > > UA1 ----- invite --port 30000-------> UA2 > > UA2 <---------OK port 20000 ---------UA2 > > > UA1 hold the call, > > UA1 -----invite --port 30000(hold) ----> UA2 > > UA1 <--------OK port 20000 -------------UA2 > > > > > > UA1 unhold the call, > > UA1 ------invite -----port 30001 -------> UA2 > UA1 <----------OK port 20000 -----------UA2 > > > > Noticed that when UA1 unhold the call, it uses a different port. > Is UA1 SIP compliant. > > Thanks again and Regards, > > TC Chan > > > On Wed, 2005-12-14 at 09:18 +0100, samuel wrote: > > UA is not SIP compliant. It must include the SDP in the OK. > > > > Samuel. > > > > 2005/12/14, tcchan <tikchoong.chan(a)redtone.com>: > > > Dear All, > > > > > > I encountered a UA that behave as follow: > > > > > > UA1 call UA2 > > > > > > UA1 ---- invite with SDP -----> UA2 > > > > > > UA1 <------OK with SDP --------UA2 > > > > > > > > > UA1 hold the call, > > > > > > UA1 ------invite (hold) ------>UA2 > > > UA1 <--------OK but not SDP ---UA2 > > > > > > > > > UA1 unhold the call > > > > > > UA1 ------invite ----------->UA2 > > > UA1 <-------OK but also no SDP---UA2 > > > > > > > > > > > > I found that most ATA(UA) would attached a SDP when reply an OK whether > > > the call is hold or unhold. But there is one ATA that does not (UA2). > > > > > > > > > I noticed that some ATA--Linksys (UA1) would send a BYE to UA2 if it > > > receive a OK with no SDP in the hold case. > > > > > > My Question is : > > > > > > Is UA2 behave according to RFC? > > > > > > > > > > > > Would deeply appreciate if someone could let me know the answer as I > > > always have trouble reading RFCs... > > > > > > > > > Thanks fot any help available. > > > > > > Regards, > > > > > > TC Chan > > > > > > > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > >

1 0

[Users] Multi Call Leg Accounting/Billing
by Lenir 15 Dec '05

15 Dec '05

Hello guys, I read in the documentation for the ACC module that it supports multi call-leg accounting via the use of AVPs, which comes in handing when doing call forwarding. However, I've googled the way with all the keywords that I could think of and I cant find an example of this feature. I understand that you have to turn it on using: modparam("acc", "multi_leg_enabled", 1) modparam("acc", "src_leg_avp_id", 110) modparam("acc", "dst_leg_avp_id", 111) But it doesn't show or say when/how to use the AVPs. Can anybody provide a sample snippet or their openser.cfg file to demonstrate how it's used? Thanks in advance! Lenir P.s. perhaps put a sample config in the documentation. The one there doesn't show how to use Multi Call-leg billing.

1 0

Re: [Serusers] simple click-to-call
by Kapil Dhawan 15 Dec '05

15 Dec '05

Well Ken gafachi service has to offer 'refer' method. No other way out. I am working with Primus and you can take minutes from them. They offer it. >From: Ken Rozinsky <xraycharlie(a)gmail.com> >To: Kapil Dhawan <sersavvy(a)hotmail.com> >Subject: Re: [Serusers] simple click-to-call >Date: Wed, 14 Dec 2005 08:49:55 -0700 > >Thanks for the help, > >I have it set up to the point where it will call me back but the >gafachi service does not support refer :( > >I'm looking around for a similar service that does. Know of a good one? > >Also, I've been trying to work around the refer by sending another >invite. The other phone will ring now after I pick up my phone but I >hear no ringing on my phone. The lines are also not joined when the >other party picks up. > >Any thoughts on how to do this using invites? > >Regards > >Ken > > > >On 12/13/05, Kapil Dhawan <sersavvy(a)hotmail.com> wrote: > > If SER is running, that means fifo is ON. > > > > > > >From: Ken Rozinsky <xraycharlie(a)gmail.com> > > >To: Kapil Dhawan <sersavvy(a)hotmail.com> > > >Subject: Re: [Serusers] simple click-to-call > > >Date: Tue, 13 Dec 2005 11:24:58 -0700 > > > > > >Thanks for the info. > > > > > >A question for you if you have the time: > > > > > >The script checks if it can write to /tmp/ser_fifo and fails. > > >The comment mention having to have the fifo server running. How do I > > >check if this is running or not? > > > > > >I see in the documantation a mention of fifo_db but I don;t think I > > >need this as am not worried about persistance. > > > > > >The permissions on /tmp/ser_fifo are currently prw-rw---- > > > > > >I tried running the script as root and get the write error also. > > > > > >Thanks again. > > > > > >Regards > > > > > >Ken > > > > > > > > >On 12/12/05, Kapil Dhawan <sersavvy(a)hotmail.com> wrote: > > > > Ken > > > > > > > > Click-to-dial is present in example directoy as ctd.sh. It takes two > > > > arguments From and To. To run it from Web-interface, you need to run > > >this > > > > command thru script. > > > > > > > > Regards > > > > > > > > > > > > >From: Ken Rozinsky <xraycharlie(a)gmail.com> > > > > >To: serusers(a)lists.iptel.org > > > > >Subject: [Serusers] simple click-to-call > > > > >Date: Mon, 12 Dec 2005 17:27:18 -0700 > > > > > > > > > >Hello, > > > > > > > > > >As an exercise in learning how all this stuff works, I'm trying to >set > > > > >up a simple click-to-call service for our office to use. And > > > > >potentialy to allow clients to use vai the web site to save them on > > > > >the long distance charges. > > > > > > > > > >I set up Asterisk with the TACI script and got it to all work with >an > > > > >account on gafachi but I noticed that it could only handle one call >at > > > > >a time. > > > > > > > > > >I was told that openSer would allow more than one call at a time so > > > > >I've installed it (basic no mySql) and have it up and running. > > > > > > > > > >Now, being new to this I have no idea how to connct to gafachi and >do > > > > >a click-to-call type thing. > > > > > > > > > >Would anybody be able to help me with this? > > > > > > > > > >Regards > > > > > > > > > >_______________________________________________ > > > > >Serusers mailing list > > > > >serusers(a)lists.iptel.org > > > > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > _________________________________________________________________ > > > > 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 > > > > _________________________________________________________________ Shah Rukh fan? Know all about the Baadshah of Bollywood. On MSN Search http://server1.msn.co.in/profile/shahrukh.asp

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[Serusers] How can I change dial rulesbased on From:' header instead of R-RURI?
by Leo Papadopoulos 14 Dec '05

14 Dec '05

Normally I would call a statement like: If (uri=^sip:123456@somewhere.com) { ..do something } I would like to implement a dial plan using the "From:" header. i.e.: If (from_uri=^sip:123456@somewhere.com) { ..do something } Anyone know if something like this exists? Leo P.

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[Users] AVPops question
by Joachim Fabini 14 Dec '05

14 Dec '05

Hi, Is there a way or workaround to generate AVPs with header values as AVP name? It seems to me that the current AVP concept defines the AVPs at OpenSER startup, so header-value-based names are not feasible. What I need is the following: Store/retrieve a (key,value) pair where the key is the current message's call-id and the value some combination of header values and pseudo- variables. e.g. use the ops avp_printf("$hdr(call-id)", "$hdr(call-id)-$Ts"); avp_pushto("$Myheader","$hdr(call-id)"); in order to append the following header to the SIP message: Myheader: 23459(a)10.0.0.1-4235627623 Any idea how to do this in OpenSER? thanks in advance --Joachim

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[Serusers] Re: Changing Sip Port
by Juan 14 Dec '05

14 Dec '05

Please disregard my previous posts. Apparently I solved the problem not by replacing the 5060 port but by adding the new port at the ser.cfg listen=10.0.1.99 port=5060 listen=10.0.1.99 port=8080 I setup the listening port and the SIP port on the Xlite to 8080 and is registering fine. ngrep is showing that the source port and the destination port both ways is 8080 so it should be working. I will test it in the real world and inform. Thanks and regards Juan

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RE: [Users] Tricky Question
by Douglas Garstang 14 Dec '05

14 Dec '05

Samuel, It seems that t_replicate() requires a host and port as parameter. It looks like it doesn't use what's in the destination set to deliver messages... which breaks stuff. Arrgh, I'm going in circles! -----Original Message----- From: samuel [mailto:samu60@gmail.com] Sent: Wednesday, December 14, 2005 10:27 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Tricky Question Have you tried t_replicate instead of t_relay??? This t_replicate function should send the REGISTER to the other registrar and absorve the answers. However this t_replicate has (or had??) an inconvenient: it can be used with only one secondary host...so it's not possible to use it with several asterisks. Please, correct me if I am wrong. Samuel. 2005/12/14, Douglas Garstang <dgarstang(a)oneeighty.com>: > Here's a tricky question for the list. > > I have phones registering with OpenSER. After authenticating and saving the location, it forwards each registration request onto multiple Asterisk systems who send back Trying, OK, etc. OpenSER forwards these back to the phones eventhough they don't need to be. I want to stop them from getting back to the phones, because from the phone's perspective they've already registered, and now these extra Trying, Ok messages are coming back. Furtunately it doesn't seem to be causing any problems for the Polycom phones, but I'd still like ton get rid of them. > > How....? Maybe call setflag() when I get a registration, and then look for messages coming back with the flag? > > Doug > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

2 1

[Users] mysql cluster and openser
by Maria Yndefors 14 Dec '05

14 Dec '05

Hallo, I run openser on a debian used the deb package to install it. I have a ndb-cluster setup that I have used successfully with SER 0.9.4, now I have tested to install openser and I can't manage to get the openser mysql module to find the correct libmysql. I have: /usr/lib/libmysqlclient.so.12 /usr/lib/libmysqlclient.so.15 I have tried with a softlink from libmysql.so.10 but I don't manage. What is the difference from standard SER and openser mysql module? Regards /Maria Maria Yndefors _______________________________________________________________ System Management & Development BREDBANDSBOLAGET / B2 Bredband AB Direct: +46 (0)8 50 69 83 80 Mobile: +46 (0)7 33 81 92 51 Fax: +46 (0)8 54 90 46 08 http://www.bredband.com _______________________________________________________________

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[Users] using TM fr_inv_timer_avp (OpenSER 1.0)
by Hilco de Lathouder 14 Dec '05

14 Dec '05

If there is an incoming request for, say, user 399(a)domain.com i want to load an alternate fr_invite_timer using AVP. According to the TM module docs, this is supported with the parameter fr_inv_timer_avp. I've studied the AVP & TM docs but i still don't fully understand how to get this working. Do I have to use avp_load statements, or does the TM module automaticaly recognize the right user ? Another question is, if i rewrite the user's URI, what will happen with the AVP? Ive created an AVP entry in the usr_preferences table for user 399(a)domain.com, with a string attribute 'hg_invite_timer' and an integer value of 5. (which would be type 1). Thanks in advance, Hilco ================= Below are my TM and AVPOPS papameters from the config file. I dont use any other avp funtion calls. # # TM params # modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_inv_timer_avp","s:hg_invite_timer") # # AVPOPS params # modparam("avpops","avp_url","mysql://openser:xxxxx@localhost/openser") modparam("avpops","avp_table","usr_preferences") =================

3 2

[Users] to terminate a parallel forking
by Matteo Piazza 14 Dec '05

14 Dec '05

Hi, If I perform a parallel forking (in a failure route after that the first call downs in timeout) I see that if one of the recever decline the call the other destination continues to ring. Is there a command to terminate all parallel calls. Matteo

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[Serusers] Netcall - Job opportunity
by Edgar Barbosa 14 Dec '05

14 Dec '05

Netcall - Job opportunity Netcall (www.netcall.pt) is the leading VoIP service provider in Portugal. Netcall is also the new kid on the block, we are fighting the established old fashioned telco's. We are currently searching for an experienced systems engineer, highly specialized on VoIP services, to join our team in Porto, Portugal. We offer you a great career opportunity in a young and dynamic company, where your contribution would have a direct impact on the business definition and strategy. You will work as a member of the development team, designing and developing next generation VoIP services. Candidates should have: - Deep knowledge on networking and VoIP related protocols (SIP, IAX, ENUM, etc.) - Experience on SER, Asterisk and MySQL - Strong troubleshooting, debugging, problem solving, and analytical skills So if you want to get for a period experience abroad and also work in a challenging area where you can express yourself with your knowledge... please make the time to address us an email with CV... If you're interested and/or would like to get additional info, please feel free to contact us: emprego(a)netcall.pt

1 0

[Serusers] PSTN guard
by Nikola Stojanoski 14 Dec '05

14 Dec '05

I'm trying to setup PSTN gateway for termination and i would like to use SER as a guard + to collect CDR's. but my consurn is the BYE messages i don't know how many failed BYE messages i'll get in realtime operation because i can't get correct cdr's without BYE. Is there a way to use mediaproxy rtp timeout so when i don't receive BYE i the call will be terminated let's say in 30 sec. What kind of message does mediaproxy send after that idle time CANCEL? And what is better way to do billing? Set cron to check balance on 5 min. or to use exec module and do mysql query after every BYE or CANCEL message? Using cron will make me lose maximum of 5 min. * number of PSTN lines and with query after every call i will lose only charge for active calls. Regards

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RE: [Users] Tricky Question
by Douglas Garstang 14 Dec '05

14 Dec '05

Samuel, I guess I can't call t_replicate more than once. Dammit! Doug. -----Original Message----- From: samuel [mailto:samu60@gmail.com] Sent: Wednesday, December 14, 2005 10:27 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Tricky Question Have you tried t_replicate instead of t_relay??? This t_replicate function should send the REGISTER to the other registrar and absorve the answers. However this t_replicate has (or had??) an inconvenient: it can be used with only one secondary host...so it's not possible to use it with several asterisks. Please, correct me if I am wrong. Samuel. 2005/12/14, Douglas Garstang <dgarstang(a)oneeighty.com>: > Here's a tricky question for the list. > > I have phones registering with OpenSER. After authenticating and saving the location, it forwards each registration request onto multiple Asterisk systems who send back Trying, OK, etc. OpenSER forwards these back to the phones eventhough they don't need to be. I want to stop them from getting back to the phones, because from the phone's perspective they've already registered, and now these extra Trying, Ok messages are coming back. Furtunately it doesn't seem to be causing any problems for the Polycom phones, but I'd still like ton get rid of them. > > How....? Maybe call setflag() when I get a registration, and then look for messages coming back with the flag? > > Doug > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[Serusers] re:Changing Sip Port
by Juan 14 Dec '05

14 Dec '05

I still can't register with another port. This is what I am getting from ngrep. interface: \ (10.0.1.0/255.255.255.0) # # U 10.0.1.105:8080 -> 10.0.1.99:8080 REGISTER sip:10.0.1.99 SIP/2.0..Via: SIP/2.0/UDP 10.0.1.105:8080;rport;bran ch=z9hG4bKA2546AEF22FA4766A172D9533D938D55..From: Juan <sip:677403479814529 @10.0.1.99>;tag=2853267163..To: Juan <sip:677403479814529@10.0.1.99>..Conta ct: "Juan" <sip:677403479814529@10.0.1.105:8080>..Call-ID: 1BEBA6527D3A4797 93E960D259872D75@10.0.1.99..CSeq: 41059 REGISTER..Expires: 1800..Max-Forwar ds: 70..User-Agent: X-Lite release 1103m..Content-Length: 0.... # I 10.0.1.99 -> 10.0.1.105 3:3 ....E...}@.........i...c........REGISTER sip:10.0.1.99 SIP/2.0..Via: SIP/2. 0/UDP 10.0.1.105:8080;rport;branch=z9hG4bKA2546AEF22FA4766A172D9533D938D55. .From: Juan <sip:677403479814529@10.0.1.99>;tag=2853267163..To: Juan <sip:6 77403479814529(a)10.0.1.99 <mailto:77403479814529@10.0.1.99>..Contact> >..Contact: "Juan" <sip:677403479814529@10.0.1.105: 8080>..Call-ID: 1BEBA6527D3A479793E960D259872D75@10.0.1.99..CSeq: 41059 REG ISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-Lite release 1103m..C ontent-Length: 0.... # U 10.0.1.105:8080 -> 10.0.1.99:8080 .. # I 10.0.1.99 -> 10.0.1.105 3:3 ....E...}B.........i...c.......... I don't know why SER (10.0.1.99) is not sending a 100 trying but instead is sending like the echo of the REGISTER request to 10.0.1.105 3:3 ? Can some one enlight me? THANKS Juan

1 0

[Serusers] Using SER together with Asterisk
by bharat.sarvan＠wipro.com 14 Dec '05

14 Dec '05

Hello all, I am trying to integrate SER and Asterisk. So could anybody please guide me with the changes and the configuration that I need to in either SER or Asterisk for achieving it. Like for suppose I want to achieve a call being placed from a User 1 registered to SER to User 2 who is registered to Asterisk.. So how do I move ahead to achieve the same. And are there any other new things that one can come up with using SER and Asterisk together which can't be achieved using SER or Asterisk alone? Kindly do let me know. Thanks and Regards, Bharat Sarvan The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[Users] Tricky Question
by Douglas Garstang 14 Dec '05

14 Dec '05

Here's a tricky question for the list. I have phones registering with OpenSER. After authenticating and saving the location, it forwards each registration request onto multiple Asterisk systems who send back Trying, OK, etc. OpenSER forwards these back to the phones eventhough they don't need to be. I want to stop them from getting back to the phones, because from the phone's perspective they've already registered, and now these extra Trying, Ok messages are coming back. Furtunately it doesn't seem to be causing any problems for the Polycom phones, but I'd still like ton get rid of them. How....? Maybe call setflag() when I get a registration, and then look for messages coming back with the flag? Doug

2 1

RE: [Users] RE: failure_route.... I could cry...
by Douglas Garstang 14 Dec '05

14 Dec '05

Klaus, Adding a second nic can't hurt. Asterisk can bind to multiple interfaces in a system. We don't have to have the info in OpenSER. We're using OpenSER as a means by which we can 'copy' the registrations to all the Asterisk systems, as Asterisk has no way to do this. By nature of the fact that OpenSER is also doing the registration authentication, it's interfacing to MySQL and for now I'm saving the location info there as well. IMO most other people's setup's look a little under-engineered. Most people seem to be happy with having a hot or even cold spare. We're a telco (CLEC) so downtown is a deal breaker. Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:41 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] RE: failure_route.... I could cry... If you already have 2 asterisk boxes, I do not think that adding a second NIC in each asterisk will increase availability. Does Asterisk bind to both interfaces? Which interface will be used by asterisk for sending requests? Which src_IP will be used? What if this interface is broken? Why do you need the location info in Asterisk and openser? IMO your setup looks like a little bit over-engineered. regards klaus Douglas Garstang wrote: > Klaus, > > Yes. It's necessary to replicate 4 times. We're implementing a 'farm' of Asterisk hosts. Asterisk has a serious limitation where there's no way to share contact info for phones between them. OpenSER seemed like it may be able to address this issue, by accepting registrations from the phones and then 'fanning' them out to ALL the Asterisk boxes. In this way, each Asterisk box knows where every phone is. Up to this point I had just been calling forward() multiple times, once to each Asterisk system and it seemed to work well. With the addition of extra ethernet ports in each Asterisk box (for extra redundancy), it got more complicated as I didn't want to register twice to the same Asterisk system. While I don't think that would cause a problem, I was just trying to make it as bullet-proof as I could. > > Doug. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 9:06 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] RE: failure_route.... I could cry... > > > That's a difficult setup. I've never did this yet and do not know if it > is possible at all. you can use t_replicate but this works only with 1 > host. I think someone solved this using forward_tcp. > > is it really necessary to replicate 4 times? > > regards > klaus > > Douglas Garstang wrote: > >>Oh dear... it's failing because as far as t_relay() is concerned, the packet has been delivered, and the transaction is complete, right? If that's the case, how can I also send the REGISTRATION packet to another destination? Can I use forward() with failure_route? Arrgh!. >> >> >> >>>-----Original Message----- >>>From: Douglas Garstang >>>Sent: Tuesday, December 13, 2005 6:01 PM >>>To: 'users(a)openser.org' >>>Subject: failure_route.... I could cry... >>> >>>I am trying to have OpenSER forward registration requests to multiple Asterisk boxes with failover. Each Asterisk box has two ip addresses. My logic tries to relay to the first IP on the first Asterisk box (192.168.10.100). If that fails, it tries to relay to the second IP on the first Asterisk box (192.168.10.17). It then goes on and repeats the process with another Asterisk box who's IP's are 192.168.10.10.101 and 192.168.10.8. >>> >>>Actually the first IP address on each Asterisk system is fake (100 and 101)... that's how I am testing that it will fail to the second one. >>> >>>The openser.cfg below is causing this to be logged to messages: >>>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route10: Asterisk-1, NIC-1 >>>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: Asterisk-2, NIC-1 >>>Dec 13 17:50:08 bil-pdev-3 openser[22318]: ERROR: t_newtran: transaction already in process 0xb6126560 < --- Why? >>>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: t_relay returned error <--- Why? >>>Dec 13 17:50:09 bil-pdev-3 openser[22328]: failure21: Failed to register with Asterisk-2, NIC-1 >>>Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: Asterisk-2, NIC-2 >>>Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: End of routine >>>Dec 13 17:50:10 bil-pdev-3 openser[22322]: User local found >>>Dec 13 17:50:10 bil-pdev-3 openser[22320]: User local found >>> >>>Why am I getting the t_newtran and t_relay errors in there? What am I doing wrong? I'm just calling rewritehostport(next-ip), followed by append_branch() and t_relay. In fact something really weird happens. When I have the code there for route[20], failure_route[21] and route[22], it causes the code for the first three routing blocks, route[10], failure_route[11] and route[12] to act differently. It seems to make route20 fail to connect eventhough it comes AFTER it in the code. >>> >>>I'd really appreciate some help! Am I doing something wrong with rewritehostport and append_branch, because quite frankly the docs are terrible. >>> >>> >>> if ( method == REGISTER ) { >>> if ( !www_authorize("voip.com", "subscriber") ) { >>> www_challenge("voip.com", "0"); >>> }; >>> save("location"); >>> route(10); >>> route(20); >>> >>># >>># First Asterisk System. >>># >>> >>># Try first IP on this system (fake - will fail) >>>route[10] { >>> xlog ("L_INFO","route10: Asterisk-1, NIC-1"); >>> t_on_failure("11"); >>> rewritehostport("192.168.10.100:5060"); >>> append_branch(); >>> if ( !t_relay() ) { >>> xlog ("L_INFO","route10: t_relay returned error"); >>> } >>>} >>> >>>failure_route[11] { >>> xlog ("L_INFO","failure11: Failed to register with Asterisk-1, NIC-1"); >>> route(12); >>>} >>> >>># Try second IP on this system (real - should work) >>>route[12] { >>> xlog ("L_INFO","route12: Asterisk-1, NIC-2"); >>> t_on_failure("13"); >>> rewritehostport("192.168.10.17:5060"); >>> append_branch(); >>> if ( !t_relay() ) { >>> xlog ("L_INFO","route12: t_relay returned error"); >>> } >>> xlog ("L_INFO","route12: End of routine"); >>> return; >>>} >>> >>># >>># Second Asterisk System >>># >>> >>># Try first IP on this system (fake - will fail) >>>route[20] { >>> xlog ("L_INFO","route20: Asterisk-2, NIC-1"); >>> t_on_failure("21"); >>> rewritehostport("192.168.10.101:5060"); >>> append_branch(); >>> if ( !t_relay() ) { >>> xlog ("L_INFO","route20: t_relay returned error");> >>> } >>>} >>> >>>failure_route[21] { >>> xlog ("L_INFO","failure21: Failed to register with Asterisk-2, NIC-1"); >>> route(22); >>>} >>> >>># Try second IP on this system (real - should work) >>>route[22] { >>> xlog ("L_INFO","route22: Asterisk-2, NIC-2"); >>> t_on_failure("23"); >>> rewritehostport("192.168.10.8:5060"); >>> append_branch(); >>> if ( !t_relay() ) { >>> xlog ("L_INFO","route22: t_relay returned error"); >>> } >>> xlog ("L_INFO","route22: End of routine"); >>> return; >>>} >>> >>>Doug >>> >>> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

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RE: [Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Let me ask this question... Does the RURI get consumed when a message is sucessfully delivered, or when a message is attempted to be delivered, but fails? -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:48 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] More Routing.... I do not know if it also works this way. I always used the version I sent in the previous email and it always worked. Thus I never had the need to change it. klaus Douglas Garstang wrote: > Ok...so... why not just call append_branch("192.168.10.8:5060") without first calling rewritehostport("192.168.10.8")? It seems like redundant information. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 9:01 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] More Routing.... > > > Where's the problem? in route[1] or in the failure route? > > You need append branch in failure route > > klaus > > Douglas Garstang wrote: > >>Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! >> >>route(1); >> >>route[1] { >> rewritehostport("192.168.10.7:5060"); >> t_on_failure("2"); >> t_relay(); >>} >> >>failure_route[2] { >> rewritehostport('192.168.10.8:5060"); >> t_relay(); >>} >> >>Doug. >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

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RE: [Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Ok...so... why not just call append_branch("192.168.10.8:5060") without first calling rewritehostport("192.168.10.8")? It seems like redundant information. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:01 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] More Routing.... Where's the problem? in route[1] or in the failure route? You need append branch in failure route klaus Douglas Garstang wrote: > Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! > > route(1); > > route[1] { > rewritehostport("192.168.10.7:5060"); > t_on_failure("2"); > t_relay(); > } > > failure_route[2] { > rewritehostport('192.168.10.8:5060"); > t_relay(); > } > > Doug. > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

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RE: [Users] RE: failure_route.... I could cry...
by Douglas Garstang 14 Dec '05

14 Dec '05

Klaus, Yes. It's necessary to replicate 4 times. We're implementing a 'farm' of Asterisk hosts. Asterisk has a serious limitation where there's no way to share contact info for phones between them. OpenSER seemed like it may be able to address this issue, by accepting registrations from the phones and then 'fanning' them out to ALL the Asterisk boxes. In this way, each Asterisk box knows where every phone is. Up to this point I had just been calling forward() multiple times, once to each Asterisk system and it seemed to work well. With the addition of extra ethernet ports in each Asterisk box (for extra redundancy), it got more complicated as I didn't want to register twice to the same Asterisk system. While I don't think that would cause a problem, I was just trying to make it as bullet-proof as I could. Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:06 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] RE: failure_route.... I could cry... That's a difficult setup. I've never did this yet and do not know if it is possible at all. you can use t_replicate but this works only with 1 host. I think someone solved this using forward_tcp. is it really necessary to replicate 4 times? regards klaus Douglas Garstang wrote: > Oh dear... it's failing because as far as t_relay() is concerned, the packet has been delivered, and the transaction is complete, right? If that's the case, how can I also send the REGISTRATION packet to another destination? Can I use forward() with failure_route? Arrgh!. > > >> -----Original Message----- >>From: Douglas Garstang >>Sent: Tuesday, December 13, 2005 6:01 PM >>To: 'users(a)openser.org' >>Subject: failure_route.... I could cry... >> >>I am trying to have OpenSER forward registration requests to multiple Asterisk boxes with failover. Each Asterisk box has two ip addresses. My logic tries to relay to the first IP on the first Asterisk box (192.168.10.100). If that fails, it tries to relay to the second IP on the first Asterisk box (192.168.10.17). It then goes on and repeats the process with another Asterisk box who's IP's are 192.168.10.10.101 and 192.168.10.8. >> >>Actually the first IP address on each Asterisk system is fake (100 and 101)... that's how I am testing that it will fail to the second one. >> >>The openser.cfg below is causing this to be logged to messages: >>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route10: Asterisk-1, NIC-1 >>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: Asterisk-2, NIC-1 >>Dec 13 17:50:08 bil-pdev-3 openser[22318]: ERROR: t_newtran: transaction already in process 0xb6126560 < --- Why? >>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: t_relay returned error <--- Why? >>Dec 13 17:50:09 bil-pdev-3 openser[22328]: failure21: Failed to register with Asterisk-2, NIC-1 >>Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: Asterisk-2, NIC-2 >>Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: End of routine >>Dec 13 17:50:10 bil-pdev-3 openser[22322]: User local found >>Dec 13 17:50:10 bil-pdev-3 openser[22320]: User local found >> >>Why am I getting the t_newtran and t_relay errors in there? What am I doing wrong? I'm just calling rewritehostport(next-ip), followed by append_branch() and t_relay. In fact something really weird happens. When I have the code there for route[20], failure_route[21] and route[22], it causes the code for the first three routing blocks, route[10], failure_route[11] and route[12] to act differently. It seems to make route20 fail to connect eventhough it comes AFTER it in the code. >> >>I'd really appreciate some help! Am I doing something wrong with rewritehostport and append_branch, because quite frankly the docs are terrible. >> >> >> if ( method == REGISTER ) { >> if ( !www_authorize("voip.com", "subscriber") ) { >> www_challenge("voip.com", "0"); >> }; >> save("location"); >> route(10); >> route(20); >> >># >># First Asterisk System. >># >> >># Try first IP on this system (fake - will fail) >>route[10] { >> xlog ("L_INFO","route10: Asterisk-1, NIC-1"); >> t_on_failure("11"); >> rewritehostport("192.168.10.100:5060"); >> append_branch(); >> if ( !t_relay() ) { >> xlog ("L_INFO","route10: t_relay returned error"); >> } >>} >> >>failure_route[11] { >> xlog ("L_INFO","failure11: Failed to register with Asterisk-1, NIC-1"); >> route(12); >>} >> >># Try second IP on this system (real - should work) >>route[12] { >> xlog ("L_INFO","route12: Asterisk-1, NIC-2"); >> t_on_failure("13"); >> rewritehostport("192.168.10.17:5060"); >> append_branch(); >> if ( !t_relay() ) { >> xlog ("L_INFO","route12: t_relay returned error"); >> } >> xlog ("L_INFO","route12: End of routine"); >> return; >>} >> >># >># Second Asterisk System >># >> >># Try first IP on this system (fake - will fail) >>route[20] { >> xlog ("L_INFO","route20: Asterisk-2, NIC-1"); >> t_on_failure("21"); >> rewritehostport("192.168.10.101:5060"); >> append_branch(); >> if ( !t_relay() ) { >> xlog ("L_INFO","route20: t_relay returned error");> >> } >>} >> >>failure_route[21] { >> xlog ("L_INFO","failure21: Failed to register with Asterisk-2, NIC-1"); >> route(22); >>} >> >># Try second IP on this system (real - should work) >>route[22] { >> xlog ("L_INFO","route22: Asterisk-2, NIC-2"); >> t_on_failure("23"); >> rewritehostport("192.168.10.8:5060"); >> append_branch(); >> if ( !t_relay() ) { >> xlog ("L_INFO","route22: t_relay returned error"); >> } >> xlog ("L_INFO","route22: End of routine"); >> return; >>} >> >>Doug >> >> > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

2 1

RE: [Users] Routing Examples
by Douglas Garstang 14 Dec '05

14 Dec '05

And you need to call append_branch().... why? The RURI is consumed? Correct? Not sure I understand when it gets consumed. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:32 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Routing Examples Then try the same without AVPs - no difference in the basic prinicples: failure_route[1] { sethostport("1.2.3.4:5061"); append_branch(); t_relay(); } Douglas Garstang wrote: > Ugh... avps....I absolutely can't follow the docs on avp's. I have -no idea- what that's all about. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 8:58 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] Routing Examples > > > Douglas Garstang wrote: > >>I'm having a terrible time trying to get failure routes to work. Can someone point me to some USEFUL examples please? The examples that come with OpenSER are trivial. They all use append_branch("sip:user@domain") which in the real world is barely useful. I need to try sending messages to a specified host with the current user's URI. >> >>For example, is this the correct usage for trying to connect in sequence to multple destinations? >> >>rewritehostport("192.168.10.7:5060"); >>append_branch(); >>t_on_failure("2"); >>t_relay(); >> >>failure_route[2] { >> rewritehostport("192.168.10.7:5060"); >> append_branch(); >> t_relay(); >>} >> >>The second route is never tried. In general, what should I be doing here? >>Which is correct? > > > > I use: > > failure_route[1] { > > # send to voicemail in format sip:<extension>@1.2.3.4:5061 > avp_pushto("$ruri/username","$extension"); > sethostport("1.2.3.4:5061"); > append_branch(); > t_relay(); > } > > klaus > > >>rewritehostport(ip-addr) >>append_branch() >>t_relay() >> >>or maybe... >>append_branch(ip-addr) >>t_relay() >> >>or maybe... >>append_branch(ip-addr1) >>append_branch(ip-addr2) >>t_relay() >> >>Do you get my point? The docs are really bad and don't cover exactly how this stuff is supposed to be implemented! if I do a google search on this stuff, I get almost no matches. There's no books either. I'm out of ideas. >> >>I'm just trying to connect to multiple destinations in sequence.... >> >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

1 0

RE: [Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Ok, I'm very confused about this. Bogdan mentioned that the RURI is consumed? When is it consumed? -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:31 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] More Routing.... append_branch will add a new branch. But the t_relay in failure route only uses the new branch, as the previous branch is alreaddy terminated. klaus Douglas Garstang wrote: > Klaus, > > But, doesn't append_branch cause parallel forking? I don't want parallel forking. If I do parallel forking, the same registration may get delivered to the same Asterisk box twice, and that's not what I want. > > Doug. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 9:01 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] More Routing.... > > > Where's the problem? in route[1] or in the failure route? > > You need append branch in failure route > > klaus > > Douglas Garstang wrote: > >>Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! >> >>route(1); >> >>route[1] { >> rewritehostport("192.168.10.7:5060"); >> t_on_failure("2"); >> t_relay(); >>} >> >>failure_route[2] { >> rewritehostport('192.168.10.8:5060"); >> t_relay(); >>} >> >>Doug. >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

1 0

RE: [Users] Routing Examples
by Douglas Garstang 14 Dec '05

14 Dec '05

Ugh... avps....I absolutely can't follow the docs on avp's. I have -no idea- what that's all about. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 8:58 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Routing Examples Douglas Garstang wrote: > I'm having a terrible time trying to get failure routes to work. Can someone point me to some USEFUL examples please? The examples that come with OpenSER are trivial. They all use append_branch("sip:user@domain") which in the real world is barely useful. I need to try sending messages to a specified host with the current user's URI. > > For example, is this the correct usage for trying to connect in sequence to multple destinations? > > rewritehostport("192.168.10.7:5060"); > append_branch(); > t_on_failure("2"); > t_relay(); > > failure_route[2] { > rewritehostport("192.168.10.7:5060"); > append_branch(); > t_relay(); > } > > The second route is never tried. In general, what should I be doing here? > Which is correct? I use: failure_route[1] { # send to voicemail in format sip:<extension>@1.2.3.4:5061 avp_pushto("$ruri/username","$extension"); sethostport("1.2.3.4:5061"); append_branch(); t_relay(); } klaus > > rewritehostport(ip-addr) > append_branch() > t_relay() > > or maybe... > append_branch(ip-addr) > t_relay() > > or maybe... > append_branch(ip-addr1) > append_branch(ip-addr2) > t_relay() > > Do you get my point? The docs are really bad and don't cover exactly how this stuff is supposed to be implemented! if I do a google search on this stuff, I get almost no matches. There's no books either. I'm out of ideas. > > I'm just trying to connect to multiple destinations in sequence.... > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

2 1

RE: [Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Klaus, But, doesn't append_branch cause parallel forking? I don't want parallel forking. If I do parallel forking, the same registration may get delivered to the same Asterisk box twice, and that's not what I want. Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:01 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] More Routing.... Where's the problem? in route[1] or in the failure route? You need append branch in failure route klaus Douglas Garstang wrote: > Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! > > route(1); > > route[1] { > rewritehostport("192.168.10.7:5060"); > t_on_failure("2"); > t_relay(); > } > > failure_route[2] { > rewritehostport('192.168.10.8:5060"); > t_relay(); > } > > Doug. > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

2 1

Re: [Users] Debug Destionation Set
by Klaus Darilion 14 Dec '05

14 Dec '05

have you tried http://openser.org/docs/pseudo-variables-1.0.x.html#pv_ds klaus Douglas Garstang wrote: > Is there a pseudo variable that will print all the members of the current destination set? A keyword search for 'destination' in the pseudo variables doc didn't find anything, so I'm guessing not. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 8:54 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] Debug Destionation Set > > > Douglas Garstang wrote: > >>Is there a way to see what's in the current destionation set? Either by a command or debug output from OpenSER. It doesn't seem to display anything in debug about the destination set at all. > > > use xlog and pseudo variables > > klaus > > >>Doug. >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

1 0

[Users] mysqlso error
by maillisting 14 Dec '05

14 Dec '05

I compile mysql.so with the following command. make modules=modules/mysql modules I have followed the document to configure openuser to use SQL. When I start openser, the error message below is shown. What does it mean the module compile flags? Any clue? openser: ERROR: module compile flags mismatch for /usr/local/lib/openser/modules/mysql.so core: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT <mailto:users@openser.org>

5 8

RE: [Users] Routing Examples
by Douglas Garstang 14 Dec '05

14 Dec '05

Bogdan... Thanks for the info. Ok... well I think I need to do serial forking. I don't want to have a situation where the registration gets delivered to two IP's on the same host because both happen to be available. I don't think Asterisk will have a problem with it, but to be safe I'd like to avoid that. So, back to my original logic of trying first ip on first Asterisk box, if that fails try second ip on first Asterisk box. Then, move onto the next and repeat the process. With pareallel forking, what happens with failures? Does a failure of any single host in destination set cause an error that can be handled with failure_route? Or, do all hosts have to fail before you can detect an error? How can you determine which one failed? I don't think parallel forking is what I need, but thanks for clearing that up for me. Doug. -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wednesday, December 14, 2005 8:55 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Routing Examples Hi Doug, if you have: rewritehostport("192.168.10.7:5060"); t_on_failure("2"); t_relay(); failure_route[2] { rewritehostport("192.168.10.7:5060"); append_branch(); t_relay(); } you do serial forking - try first destination and if fails, o for the second one. for: rewritehostport("192.168.10.7:5060"); append_branch("192.168.10.7:5060"); t_relay() you do parallel forking - the request will be sent in the same time to all destinations. regards, bogdan Douglas Garstang wrote: >I'm having a terrible time trying to get failure routes to work. Can someone point me to some USEFUL examples please? The examples that come with OpenSER are trivial. They all use append_branch("sip:user@domain") which in the real world is barely useful. I need to try sending messages to a specified host with the current user's URI. > >For example, is this the correct usage for trying to connect in sequence to multple destinations? > >rewritehostport("192.168.10.7:5060"); >append_branch(); >t_on_failure("2"); >t_relay(); > >failure_route[2] { > rewritehostport("192.168.10.7:5060"); > append_branch(); > t_relay(); >} > >The second route is never tried. In general, what should I be doing here? >Which is correct? > >rewritehostport(ip-addr) >append_branch() >t_relay() > >or maybe... >append_branch(ip-addr) >t_relay() > >or maybe... >append_branch(ip-addr1) >append_branch(ip-addr2) >t_relay() > >Do you get my point? The docs are really bad and don't cover exactly how this stuff is supposed to be implemented! if I do a google search on this stuff, I get almost no matches. There's no books either. I'm out of ideas. > >I'm just trying to connect to multiple destinations in sequence.... > > > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

1 0

Re: [Users] More Routing....
by Bogdan-Andrei Iancu 14 Dec '05

14 Dec '05

Doug, in failure route, the RURI was already consumed (you already got reply for it) - so, if you want to relay again, you need to append a new branch. regards, bogdan Douglas Garstang wrote: >Bogdan, > >I don't understand why I need to do this. Doesn't rewritehostport rewrite the current URI so that t_relay can statefully deliver? If so, what's the point of append_branch? It seems redundant, which is why I don't understand the need for it. > >Doug. > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Wednesday, December 14, 2005 8:47 AM >To: Douglas Garstang >Cc: users(a)openser.org >Subject: Re: [Users] More Routing.... > > >Hi Doug, > >in failure route, in order to fork a new serial branch, you need to call >append_branch() before t_relay(). > >regards, >bogdan > >Douglas Garstang wrote: > > > >>Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! >> >>route(1); >> >>route[1] { >> rewritehostport("192.168.10.7:5060"); >> t_on_failure("2"); >> t_relay(); >>} >> >>failure_route[2] { >> rewritehostport('192.168.10.8:5060"); >> t_relay(); >>} >> >>Doug. >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> > > >

1 0

[Serusers] problems with RTP proxy
by Anthony Chapellier 14 Dec '05

14 Dec '05

Hi, I've got troubles with RTP proxy. When I initiate a call between two UAs, I see RTP packets arriving from both clients to RTP proxy but they don't leave the proxy and of course clients don't receive them... does someone know what the problem could be ? Thanks,

2 1

[Users] RE: failure_route.... I could cry...
by Douglas Garstang 14 Dec '05

14 Dec '05

Oh dear... it's failing because as far as t_relay() is concerned, the packet has been delivered, and the transaction is complete, right? If that's the case, how can I also send the REGISTRATION packet to another destination? Can I use forward() with failure_route? Arrgh!. > -----Original Message----- > From: Douglas Garstang > Sent: Tuesday, December 13, 2005 6:01 PM > To: 'users(a)openser.org' > Subject: failure_route.... I could cry... > > I am trying to have OpenSER forward registration requests to multiple Asterisk boxes with failover. Each Asterisk box has two ip addresses. My logic tries to relay to the first IP on the first Asterisk box (192.168.10.100). If that fails, it tries to relay to the second IP on the first Asterisk box (192.168.10.17). It then goes on and repeats the process with another Asterisk box who's IP's are 192.168.10.10.101 and 192.168.10.8. > > Actually the first IP address on each Asterisk system is fake (100 and 101)... that's how I am testing that it will fail to the second one. > > The openser.cfg below is causing this to be logged to messages: > Dec 13 17:50:08 bil-pdev-3 openser[22318]: route10: Asterisk-1, NIC-1 > Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: Asterisk-2, NIC-1 > Dec 13 17:50:08 bil-pdev-3 openser[22318]: ERROR: t_newtran: transaction already in process 0xb6126560 < --- Why? > Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: t_relay returned error <--- Why? > Dec 13 17:50:09 bil-pdev-3 openser[22328]: failure21: Failed to register with Asterisk-2, NIC-1 > Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: Asterisk-2, NIC-2 > Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: End of routine > Dec 13 17:50:10 bil-pdev-3 openser[22322]: User local found > Dec 13 17:50:10 bil-pdev-3 openser[22320]: User local found > > Why am I getting the t_newtran and t_relay errors in there? What am I doing wrong? I'm just calling rewritehostport(next-ip), followed by append_branch() and t_relay. In fact something really weird happens. When I have the code there for route[20], failure_route[21] and route[22], it causes the code for the first three routing blocks, route[10], failure_route[11] and route[12] to act differently. It seems to make route20 fail to connect eventhough it comes AFTER it in the code. > > I'd really appreciate some help! Am I doing something wrong with rewritehostport and append_branch, because quite frankly the docs are terrible. > > > if ( method == REGISTER ) { > if ( !www_authorize("voip.com", "subscriber") ) { > www_challenge("voip.com", "0"); > }; > save("location"); > route(10); > route(20); > > # > # First Asterisk System. > # > > # Try first IP on this system (fake - will fail) > route[10] { > xlog ("L_INFO","route10: Asterisk-1, NIC-1"); > t_on_failure("11"); > rewritehostport("192.168.10.100:5060"); > append_branch(); > if ( !t_relay() ) { > xlog ("L_INFO","route10: t_relay returned error"); > } > } > > failure_route[11] { > xlog ("L_INFO","failure11: Failed to register with Asterisk-1, NIC-1"); > route(12); > } > > # Try second IP on this system (real - should work) > route[12] { > xlog ("L_INFO","route12: Asterisk-1, NIC-2"); > t_on_failure("13"); > rewritehostport("192.168.10.17:5060"); > append_branch(); > if ( !t_relay() ) { > xlog ("L_INFO","route12: t_relay returned error"); > } > xlog ("L_INFO","route12: End of routine"); > return; > } > > # > # Second Asterisk System > # > > # Try first IP on this system (fake - will fail) > route[20] { > xlog ("L_INFO","route20: Asterisk-2, NIC-1"); > t_on_failure("21"); > rewritehostport("192.168.10.101:5060"); > append_branch(); > if ( !t_relay() ) { > xlog ("L_INFO","route20: t_relay returned error");> > } > } > > failure_route[21] { > xlog ("L_INFO","failure21: Failed to register with Asterisk-2, NIC-1"); > route(22); > } > > # Try second IP on this system (real - should work) > route[22] { > xlog ("L_INFO","route22: Asterisk-2, NIC-2"); > t_on_failure("23"); > rewritehostport("192.168.10.8:5060"); > append_branch(); > if ( !t_relay() ) { > xlog ("L_INFO","route22: t_relay returned error"); > } > xlog ("L_INFO","route22: End of routine"); > return; > } > > Doug > >

3 2

[Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! route(1); route[1] { rewritehostport("192.168.10.7:5060"); t_on_failure("2"); t_relay(); } failure_route[2] { rewritehostport('192.168.10.8:5060"); t_relay(); } Doug.

3 2

[Users] Routing Examples
by Douglas Garstang 14 Dec '05

14 Dec '05

I'm having a terrible time trying to get failure routes to work. Can someone point me to some USEFUL examples please? The examples that come with OpenSER are trivial. They all use append_branch("sip:user@domain") which in the real world is barely useful. I need to try sending messages to a specified host with the current user's URI. For example, is this the correct usage for trying to connect in sequence to multple destinations? rewritehostport("192.168.10.7:5060"); append_branch(); t_on_failure("2"); t_relay(); failure_route[2] { rewritehostport("192.168.10.7:5060"); append_branch(); t_relay(); } The second route is never tried. In general, what should I be doing here? Which is correct? rewritehostport(ip-addr) append_branch() t_relay() or maybe... append_branch(ip-addr) t_relay() or maybe... append_branch(ip-addr1) append_branch(ip-addr2) t_relay() Do you get my point? The docs are really bad and don't cover exactly how this stuff is supposed to be implemented! if I do a google search on this stuff, I get almost no matches. There's no books either. I'm out of ideas. I'm just trying to connect to multiple destinations in sequence....

3 2

[Users] Debug Destionation Set
by Douglas Garstang 14 Dec '05

14 Dec '05

Is there a way to see what's in the current destionation set? Either by a command or debug output from OpenSER. It doesn't seem to display anything in debug about the destination set at all. Doug.

3 2

[Users] connection failed
by maillisting 14 Dec '05

14 Dec '05

Hi all, Anyone knows the meaning of the following errors? Anyway to solve it? Dec 9 17:16:00 ow01 ./openser[19379]: ERROR: tcp_blocking_connect: timeout (10) Dec 9 17:16:00 ow01 ./openser[19379]: ERROR: tcpconn_connect: tcp_blocking_connect failed Dec 9 17:16:00 ow01 ./openser[19379]: ERROR: tcp_send: connect failed Dec 9 17:16:00 ow01 ./openser[19379]: msg_send: ERROR: tcp_send failed Dec 9 17:16:00 ow01 ./openser[19379]: ERROR:tm:t_forward_nonack: sending request failed Dec 9 17:16:00 ow01 ./openser[19379]: ERROR:tm: _reply_light: can't generate 477 reply when a final 487 was sent out Dec 9 17:16:00 ow01 ./openser[19379]: ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL)

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[Users] Use of append_branch
by Douglas Garstang 14 Dec '05

14 Dec '05

How can I call append branch and use a variable name in the user port (ie the user for which this message is addressed) of the URI? _ALL_ the examples have something like: append_branch("sip:nonsense@192.168.2.16:5064"); which is completely useless. I need to call append_branch and have the user portion of the uri inserted. Something like: append_branch("sip:$to_user@192.168.2.16:5064"); How can I do this? Doug.

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Re: [Devel] [Users] nathelper/rtpproxy when both SIP UA are behind same NAT
by Norman Brandinger 14 Dec '05

14 Dec '05

Hi Rafael, You would use the "example code snippet" when you want to decide whether or not to use rtpproxy or mediaproxy based on the fact that both the caller and the callee are behind the same NAT device. Please review the mailing lists as this topic has been discussed many times. I added some additional comments in regard to the setting of flags 2 and 3 in the posted example at: http://openser.org/dokuwiki/doku.php?id=avp_examples Generally, I have found that if OpenSER fails to start with an error indicating that the function can't be found, it's time to find the correct module to add. In response to your last question "etc." I suggest that you review the three web sites: openser.org, iptel.org and onsip.org. The last one has some really great material that explains alot. Regards, Norm Rafael R. GV wrote: > Hi > I am using nathelper/rtpproxy in ser , please see my ser.cfg attached > and tell me where do I have to use this code snippet? where did you > set flags 2 and 3?, what other modules I need?, etc. > > thank you > Rafael > Lima-Peru > > On 12/7/05, *Norman Brandinger* <norm(a)goes.com <mailto:norm@goes.com>> > wrote: > > Thanks go out to Klaus and Tavis. > > I took the results of this thread and placed it in the docuwiki > for the > rest of the user community to use (at least, the rest of the user > community that reads the docuwiki :) > > http://openser.org/dokuwiki/doku.php?id=avp_examples > > If I made any typos, please feel free to correct them. > > Regards, > Norm > > _______________________________________________ > Users mailing list > Users(a)openser.org <mailto:Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users > <http://openser.org/cgi-bin/mailman/listinfo/users> > > > ------------------------------------------------------------------------ > > # > # If you don't want to enforce RTP proxy for some destinations > # then simply use t_relay() instead of route(1) > # > # Modificado para MYSQL_ACC support > # Handling of Unavailable user and Voicemail redirection (404|408|486) > # > # Log Missed Calls (403|404|408|415|484|486|487). > # > # Group checking and PSTN credentials. > # > # 18/07/2005 Prepaid & B2bua support added > # 11/11/2005 LDI a 192.168.2.132 with a2billing > # > # Pendientes !! - Administrar Multi-Dominio para otros Códigos de Area. > # - Completar Logica para tratamiento de llamadas entre > # un mismo NAT usando avpops. > # > # ----------- global configuration parameters ------------------------ > > #/* Uncomment these lines to enter debugging mode > debug=2 > fork=yes > log_stderror=yes > #*/ > > listen=192.168.2.130 > listen=127.0.0.1 > port=5060 > > # hostname matching an alias will satisfy the condition uri==myself". > alias=mydomain.com.pe > alias=127.0.0.1 > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > children=4 > fifo="/tmp/ser_fifo" > fifo_mode=0666 # Fifo permissions can be changes from here. > > # sip_warning - Should replies include extensive warnings? > # By default yes, it is good for trouble-shooting. > sip_warning=yes > > # ------------------ module loading ---------------------------------- > loadmodule "/usr/local/lib/ser/modules/domain.so" > loadmodule "/usr/local/lib/ser/modules/avpops.so" > loadmodule "/usr/local/lib/ser/modules/mysql.so" > loadmodule "/usr/local/lib/ser/modules/sl.so" > loadmodule "/usr/local/lib/ser/modules/tm.so" > loadmodule "/usr/local/lib/ser/modules/rr.so" > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > loadmodule "/usr/local/lib/ser/modules/usrloc.so" > loadmodule "/usr/local/lib/ser/modules/registrar.so" > loadmodule "/usr/local/lib/ser/modules/group.so" > loadmodule "/usr/local/lib/ser/modules/uri.so" > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > loadmodule "/usr/local/lib/ser/modules/acc.so" > loadmodule "/usr/local/lib/ser/modules/textops.so" > > # digest authentication > loadmodule "/usr/local/lib/ser/modules/auth.so" > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > # !! Nathelper > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > # ----------------- setting module-specific parameters --------------- > > modparam("usrloc", "db_mode", 2) > > # minimize write back window - default is 60 seconds > modparam("usrloc", "timer_interval", 10) > > # database location > modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") > > modparam("usrloc", "use_domain", 1) > modparam("auth_db", "use_domain", 1) > > > modparam("domain", "db_mode", 1) > modparam("domain", "domain_table", "domain") > modparam("domain", "domain_col", "domain") > > > # ------------- Mysql Accounting parameters > modparam("acc", "log_flag", 1) > modparam("acc", "log_level", 2) > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 3) > modparam("acc", "log_missed_flag", 3) > modparam("acc", "db_url", "mysql://seradmin:heslo@localhost/ser") > modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc (para INVITE y ACK) > modparam("acc", "failed_transactions", 1) # *all* non-200 transactions marked for acc will be logged too. > > modparam("acc", "log_fmt", "miocfsputdr") > > modparam("tm", "fr_timer", 20 ) > modparam("tm", "fr_inv_timer", 40 ) # Timer which hits if no final reply for an INVITE > modparam("tm", "wt_timer", 20 ) > > # add value to ;lr param to make some broken UAs happy > modparam("rr", "enable_full_lr", 1) > > modparam("group", "db_url", "mysql://seradmin:heslo@localhost/ser") > modparam("uri_db", "db_url", "mysql://seradmin:heslo@localhost/ser") > > # ------------- registration parameters > modparam("registrar", "nat_flag", 6) > modparam("registrar", "min_expires", 60) > modparam("registrar", "max_expires", 86400) > modparam("registrar", "default_expires", 3600) > modparam("registrar", "desc_time_order", 1) > modparam("registrar", "append_branches", 1) > > modparam("registrar", "use_domain", 1) > > # !! Nathelper > # modparam("registrar", "nat_flag", 6) > modparam("nathelper", "natping_interval", 30) # Ping interval 30 s > modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT > > # -------------------------- request routing logic -------------------------- > > route { > > log(1, "-------------------------------------------\n"); > log(1, "entering main loop\n"); > > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > break; > }; > if ( msg:len >= max_len ) { > sl_send_reply("513", "Message too big"); > break; > }; > > # !! Nathelper > # Special handling for NATed clients; first, NAT test is > # executed: it looks for via!=received and RFC1918 addresses > # in Contact (may fail if line-folding is used); also, > # the received test should, if completed, should check all > # vias for rpesence of received > if (nat_uac_test("19")) { > # Allow RR-ed requests, as these may indicate that > # a NAT-enabled proxy takes care of it; unless it is > # a REGISTER > > if (method == "REGISTER" || ! search("^Record-Route:")) { > log("LOG: Someone trying to register from private IP, rewriting\n"); > > # This will work only for user agents that support symmetric > # communication. We tested quite many of them and majority is > # smart enough to be symmetric. In some phones it takes a configuration > # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is > # called "symmetric media" and "symmetric signalling". > > fix_nated_contact(); # Rewrite contact with source IP of signalling > if (method == "INVITE") { > fix_nated_sdp("1"); # Add direction=active to SDP > }; > force_rport(); # Add rport parameter to topmost Via > setflag(6); # Mark as NATed > }; > }; > > > # set flag for Radius Accounting: > > > if (!method=="OPTIONS") setflag(3); # SET MISSED_CALLS FLAG FOR ACC > > if (method=="INVITE") { > log(1, "INVITE MESSAGE RECEIVED - START ACC\n"); > setflag(1); /* set for accounting (the same value as in log_flag!) */ > }; > > if (method=="BYE") { > log (1, "BYE - STOP ACCOUNTING\n"); > setflag(1); > }; > > if (method=="CANCEL") { > log (1, "CANCEL - STOP ACCOUNTING\n"); > setflag(1); > }; > > > # 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 > # Excluding packets from b2bua (port 5070) > if (loose_route() ) { > log(1," Mark routing logic in request --> rr-enforced \n"); > append_hf("P-hint: rr-enforced\r\n"); > # t_relay(); > route(1); # Nathelper!! > break; > }; > > if (!uri==myself ) { > log(1," Mark routing logic in request --> outbound\r \n"); > append_hf("P-hint: outbound\r\n"); > # t_relay(); > route(1); # Nathelper!! > break; > }; > > if (uri==myself) { > > if (method == "REGISTER") { > > log(1, "ANALYZING REGISTER REQUEST\n"); > # to use digest authentication > if (is_user_in("Request-URI", "desactivado")) { > sl_send_reply("402", "Su cuenta fue desactivada"); > break; > }; > > if (!www_authorize("mydomain.com.pe", "subscriber")) { > log(1," ----- Fails to Register \n"); > www_challenge("mydomain.com.pe", "0"); > break; > }; > > # only signed users are allowed > if (!check_to()) { > log(1, "LOG: Hijack!!!--> unsigned user registration attempt\n"); > sl_send_reply("403", "hijack attempt!!!! Only signed users are allowed"); > break; > }; > log(1," Registered!!! \n"); > if (!save("location")) { > sl_reply_error(); > }; > break; > }; > > > #For *B2bua: First check the source of the call > #********************************************** > # If the call comes from the gateways, no authentication is required. > if (src_ip==192.168.2.145 || src_ip==192.168.2.132 || src_ip==192.168.2.131) { > log(1,"Call from pstn|*, no authentication is required. \n"); > > # If the call comes from B2BUA, no authentication is required, > # The first leg of the call has already been authenticated. > } else if (src_ip==192.168.2.130 && src_port==5070) { > log(1,"Call from B2BUA, no authentication is required. \n"); > } else { > # We check user credentials > if ((method == "INVITE" || method== "CANCEL" || method== "BYE" || method== "ACK") && (!src_ip==192.168.2.130 && !src_port==5070)){ > log(1, "ANALYZING INVITE||CANCEL REQUESTs\n"); > if (!proxy_authorize("mydomain.com.pe", "subscriber")) { > # log(1," ----- Fails to ...proxy_authorize \n"); > proxy_challenge("mydomain.com.pe", "0"); > break; > } else { > if (method == "INVITE" && !check_from()) { > sl_send_reply("403", "Only registered users are allowed"); > log(1," ----> Only registered users are allowed \n"); > break; > }; > }; > > # Not all the users are PREPAID, so we check the database > # to see if the call will be routed through B2BUA. > # If every call is to be routed through B2BUA the "is_user_in" > # conditional is not required. > # Do not use b2bua for local calls > > if (is_user_in("From", "prepaidb") && uri=~"^sip:00") { > log(1," ----> Usuario PREPAGO!!! enviando a b2bua... \n"); > rewritehostport("192.168.2.130:5070"); > t_relay_to_udp("192.168.2.130", "5070"); > break; > }; > > };# End of if (method == "INVITE" |... > }; > > > /* *********** Dial out to Local and PSTN logic ********* */ > > > # Forward +9n digit requests to gateway Cisco-AS5350 (Celulares): > > if(uri=~"^sip:9" ){ > log(1," digit expression match - Celulares \n"); > if (!is_user_in("from", "movlim")) { > sl_send_reply("403", "No permission for mobile calls"); > acc_db_request("403 Forbidden", "missed_calls"); > break; > }; > rewritehostport("192.168.2.145:5060"); > route(1); > break; > }; > > > /* ******************************************************************** */ > > lookup("aliases"); > > > # does the user wish redirection on no availability? (i.e., is he > # in the voicemail group?) -- determine it now and store it in > # flag 4, before we rewrite the flag using UsrLoc > > if (is_user_in("Request-URI", "voicemail")) { > log(1, "requested user is in voicemail group \n"); > setflag(4); > }; > > > # native SIP destinations are handled using our USRLOC DB: > > # LOOKUP (location)!!!! > > if (!lookup("location")) { > log(1,"unable to locate user X ... sending to route(4)! \n"); > # handle user which was not found > route(4); > break; > }; > > ### Test if UAS are in the same NAT: > > # get the host part of the final uri (IP part) and store it in AVP ID 13 > > avp_write("$ruri/domain", "i:13"); > if (avp_check("i:13","eq/$src_ip/i")) { > log(1, "source IP is the same as destination IP\n"); > route(3); > break; > }; > avp_delete("i:13/g"); > > > }; # End of "if(uri==myself)" > > > append_hf("P-hint: usrloc applied\r\n"); > route(1); > > # if user is on-line and is in Voicemail group, enable redirection > if (method == "INVITE" && isflagset(4)) { > log(1, "invite for voicemail user->initiate failureroute[1]\n"); > t_on_failure("1"); > }; > } > > ### ##### ####### ########## - ROUTES - ############### ################# ################## > > route[1] > { > > # if client or server know to be behind a NAT, enable relay > if (isflagset(6)) { > log(1, "Route1: force rtp proxy!!!\n"); > force_rtp_proxy(); > }; > > # NAT processing of replies; apply to all transactions (for example, > # re-INVITEs from public to private UA are hard to identify as > # NATed at the moment of request processing); look at replies > t_on_reply("1"); > > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > if (!t_relay()) { > sl_reply_error(); > }; > log(1, "Route[1]: Send it out now!!!\n"); > break; > } > > # !! Nathelper > onreply_route[1] { > # NATed transaction ? > # Not all 2xx messages have a content body so here we > # make sure our Content-Length > 0 to avoid a parse error > > if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { > fix_nated_contact(); > if (!search("^Content-Length:\ 0")) { > force_rtp_proxy(); > }; > # otherwise, is it a transaction behind a NAT and we did not > # know at time of request processing ? (RFC1918 contacts) > } else if (nat_uac_test("1")) { > fix_nated_contact(); > }; > } > > # -------------- SIP-to-PSTN call routed --------------------- > > route[2]{ > log(1,"route[2]:SIP-to-GW call routed \n"); > if(!t_relay()){ > sl_reply_error(); > }; > log(1, "Route[2]: Send it out now!!!\n"); > } > > # -------------- Same NAT Call Routing (no force rtpproxy) ---- > > route[3]{ > log(1," route[3]: UAs are in the same nat, NO force_rtp_proxy "); > > # What do I have to do here? > > if(!t_relay()){ > sl_reply_error(); > }; > log(1, "Route[3]: Send it out now!!!\n"); > } > > # --------------- Handling of Unavailable user ---------------- > route[4] { > > # non-Voip -- just send "off-line" > if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" || method=="BYE" || method=="OPTIONS")) { > sl_send_reply("404", "Not Found"); > acc_db_request("404 Not Found", "missed_calls"); > log(1, "acc 404 Not Found 1 \n"); > break; > }; > > if (!isflagset(4) && !method=="OPTIONS" && !method=="ACK" && !method=="BYE") { > sl_send_reply("404", "Not Found and no voicemail turned on !! "); > acc_db_request("404 Not Found", "missed_calls"); > log(1, " acc 404 Not Found and no voicemail \n"); > break; > }; > > # forward to voicemail adding prefix to simplify asterisk "extension.conf" > prefix("vm"); > acc_db_request("404 Not Found -> Vm", "missed_calls"); > rewritehostport("192.168.2.131:5070"); > t_relay_to_udp("192.168.2.131", "5070"); > } > > # if forwarding downstream did not succeed, try voicemail running at Asterisk > > failure_route[1]{ > if (t_check_status("408")){ > # revert_uri (); back to the original URI, makes me loose all lookup/rewrite stuff > prefix("vm"); > rewritehostport ("192.168.2.131:5070"); > acc_db_request("408 Timeout -> Vm", "missed_calls"); > append_branch(); > t_relay(); > break; > } else if (t_check_status("486")){ > # revert_uri (); back to the original URI, makes me loose all lookup/rewrite stuff > prefix("vm"); > rewritehostport ("192.168.2.131:5070"); > acc_db_request("486 Busy -> Vm", "missed_calls"); > append_branch(); > t_relay(); > break; > } > } > > ### The End ###

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Re: [Serusers] Changing Sip Port
by Voipers Portugal 14 Dec '05

14 Dec '05

There are 2 port setting that you can change. Either you change the signalling port to contact SER: when specifying the SIP Server just do: sip_server:port In this case, if you change the port, make sure that your SER Server is listening on that port as well. Or you can change the listen port of your Softphone. In this case, you don't need to alter anything else. Not sure about what you wanted. Can you provide the SIP messages log? Jose Simoes On 12/13/05, Juan <juan(a)uwtcallback.com> wrote: > > > Hi Jose. > I tried your way but I did not get it registred. I used many ports but > couldn't get it registred. > I am not using any authentication and when using port 5060 the softphone > registres OK. > My SER and my softphone (X- lite) are on the same network. > Any ideas? > Thanks in advance, > > Juan Ferrari > United World Telecom > > > ------------------------------ > *From:* serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] *On > Behalf Of *Voipers Portugal > *Sent:* Saturday, December 10, 2005 8:48 AM > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Changing Sip Port > > > Most softphones allow that. When you are specifying a SIP server, for > default (normally) it uses 5060. If you want to alter this port just do the > following: > > ser_IP_address:port_number > > 127.0.0.1:5090 > > Jose Simoes > > p.s. this works with X-Lite > > On 12/5/05, Andrey Kouprianov <andrey.kouprianov(a)gmail.com> wrote: > > > > I guess it depends on your softphone... if it chooses to communicate > > from port 5060, there's nothing you can do about it, unless you can > > change it in the phone's settings... > > Some softphones, for instance, get a random port... > > > > On 12/5/05, sagar <vids_cs(a)yahoo.com> wrote: > > > Hi All, > > > > > > I am using ser 0.8.14 +asterisk B2bua. > > > > > > Currently softphone is registering on standard sip port 5060 . > > > Is it possibe to use any other port like 5090 or 8069 > > > > > > > > > I tried to do so by changing the listining port to 5090 in ser.cfgbut in > > > case the softphone is not getting register. > > > > > > > > > Here is the ser log > > > > > > > > > 0(11348) SIP Request: > > > 0(11348) method: <REGISTER> > > > 0(11348) uri: <sip: 192.168.1.100> > > > 0(11348) version: <SIP/2.0> > > > 0(11348) parse_headers: flags=1 > > > 0(11348) end of header reached, state=5 > > > 0(11348) parse_headers: Via found, flags=1 > > > 0(11348) parse_headers: this is the first via > > > 0(11348) After parse_msg... > > > 0(11348) preparing to run routing scripts... > > > 0(11348) DEBUG : is_maxfwd_present: searchi ng for max_forwards > > header > > > 0(11348) parse_headers: flags=128 > > > 0(11348) DEBUG: is_maxfwd_present: value = 70 > > > 0(11348) parse_headers: flags=8 > > > 0(11348) DEBUG: add_param: > > > tag=fa37b062306344d68023aed26289e820 > > > 0(11348) DEBUG: add_param: epid=f657995ecc > > > 0(11348) end of header reached, state=29 > > > 0(11348) parse_headers: flags=256 > > > 0(11348) end of header reached, state=9 > > > 0(11348) DEBUG: get_hdr_field: <To> [37]; > > > uri=[sip:2851816416@192.168.1.100:5090] > > > 0(11348) DEBUG: to body [<sip:2851816416@192.168.1.100:5090> > > > ] > > > 0(11348) get_hdr_field: cseq <CSeq>: <1> <REGISTER> > > > 0(11348) DEBUG: get_hdr_body : content_length=0 > > > 0(11348) found end of header > > > 0(11348) find_first_route(): No Route headers found > > > 0(11348) loose_route(): There is no Route HF > > > 0(11348) parse_headers: flags=64 > > > 0(11348) parse_headers: flags=64 > > > 0(11348) parse_headers: flags=33554432 > > > 0(11348) check_self - checking if host==us: 13==13 && [ > > 192.168.1.100] == > > > [192.168.1.100 ] > > > 0(11348) check_self - checking if port 5090 matches port 5060 > > > 0(11348) check_self: host != me > > > 0(11348) DEBUG: t_addifnew: msg id=1 , global msg id=0 , T on > > > entrance=0xffffffff > > > 0(11348) parse_headers: flags=-1 > > > 0(11348) parse_headers: flags=60 > > > 0(11348) t_lookup_request: start searching: hash=58712, isACK=0 > > > 0(11348) DEBUG: proceeding to pre-RFC3261 transaction matching > > > 0(11348) DEBUG: t_lookup_request: no transaction found > > > 0(11348) DEBUG: mk_proxy: doing DNS lookup... > > > 0(11348) check_via_address(192.168.1.2, 192.168.1.2, 0) > > > 0(11348) DEBUG: add_to_tail_of_timer[4]: 0xb6165fe8 > > > 0(11348) DEBUG: add_to_tail_of_timer[0]: 0xb6165ffc > > > 0(11348) SER: new transacti on fwd'ed > > > 0(11348) DEBUG:destroy_avp_list: destroing list (nil) > > > 0(11348) receive_msg: cleaning up > > > > > > Please advise. > > > > > > Thanks In Advance. > > > > > > > > > ________________________________ > > > Yahoo! Personals > > > Single? There's someone we'd like you to meet. > > > Lots of someones, actually. Yahoo! Personals > > > > > > > > > _______________________________________________ > > > 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 > > > >

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Re: [Serusers] Please, help with SUBSCRIBE timeouts
by Andrey Kouprianov 14 Dec '05

14 Dec '05

yes.. thanx On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > I know such situations from my work too. If the problems show again let me > know... > Vaclav > > On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > > > Thanx a lot for understanding. :)))) > > > > On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > >> :-) It happens... > >> Vaclav > >> > >> On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > >> > >>> And then again my UA is working OK now... Im just stunned... I have a > >>> deadline for this app. Have to present it on coming friday... Hehe... > >>> If it fails to SUBSCRIBE... I'll be in trouble... > >>> > >>> On 12/14/05, Andrey Kouprianov <andrey.kouprianov(a)gmail.com> wrote: > >>>> Yes, i've noticed that too... And yes my config implies statefull > >>>> forwarding.... I think it's my mistake... I was trying with forward > >>>> instead of t_relay at the time I generated traces... > >>>> > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >>> > >> > > >

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[Serusers] using pa module
by lo ™ 14 Dec '05

14 Dec '05

hi all, i tried to use the presence module "pa.so" (without jabber support or xcap authorization) but i don't understand which functionalities have been fixed and which are currently unstable... i'm using eyebeam and i follow these "Configuring Presence Agent" instructions: http://pals.internet2.edu/trials/pic-ser/eyeBeamInstall.html this is part of my ser.cfg: if (method=="PUBLISH") { if (!t_newtran()) { log(1,"ser.cfg: PUBLISH: t_newtran error\n"); sl_reply_error(); }; log(1,"ser.cfg: PUBLISH: handle_publish\n"); handle_publish("registrar"); break; } else if (method=="SUBSCRIBE") { if (!t_newtran()) { log(1,"ser.cfg: SUBSCRIBE: newtran error\n"); sl_reply_error(); }; log(1,"ser.cfg: SUBSCRIBE: handle_subscription\n"); handle_subscription("registrar"); break; }; is it correct? i would like to see my contacts to eventually change state... is it possible? thank you!

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[Serusers] SIP call hold/unhold
by tcchan 14 Dec '05

14 Dec '05

Dear All, I encountered a UA that behave as follow: UA1 call UA2 UA1 ---- invite with SDP -----> UA2 UA1 <------OK with SDP --------UA2 UA1 hold the call, UA1 ------invite (hold) ------>UA2 UA1 <--------OK but not SDP ---UA2 UA1 unhold the call UA1 ------invite ----------->UA2 UA1 <-------OK but also no SDP---UA2 I found that most ATA(UA) would attached a SDP when reply an OK whether the call is hold or unhold. But there is one ATA that does not (UA2). I noticed that some ATA--Linksys (UA1) would send a BYE to UA2 if it receive a OK with no SDP in the hold case. My Question is : Is UA2 behave according to RFC? Would deeply appreciate if someone could let me know the answer as I always have trouble reading RFCs... Thanks fot any help available. Regards, TC Chan

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Re: [Serusers] end the session if no media-conclusion
by Hakan YASTI 14 Dec '05

14 Dec '05

Dear All, After studied on different scenarios, we can not end the session when there is no rtp on media channels. I have heard that some media proxies can do this (like nextone,acmepacket) but I did not test it. As a brief, if the bye will not come from one of the legs because of network problems, you do not have chance to end the session by checking the rtp packets. How did we solve the problem ? Now we are using a session controller. when a session starts, we start a timer, after 1 hour, if the session is going on we are sending a notify about thr session to the network control center. They can kill the session through the session controller. In any case, after 2 hours later if the session is again going on, we kill it automatically from the session controller and send a notification mail to the sysadmin which contains the session information. In this scenario, my customers do not have sessions more than 2 hours. ----- Original Message ----- From: Greger V. Teigre To: Hakan YASTI Sent: Thursday, November 17, 2005 9:50 AM Subject: Re: [Serusers] end the session if no media Good luck :-) And please copy serusers on your conclusions. g-) ----- Original Message ----- From: Hakan YASTI To: Greger V. Teigre Sent: Wednesday, November 16, 2005 1:46 PM Subject: Re: [Serusers] end the session if no media Dear Greger, sorry for late reply. I was on vacation. The scenario is like that. A ( ATA ) ----------------> send INVITE to SER ------------> SER INVITES to session controller --------> session controller invites to ---------> soft switch --------------call setup to ------PSTN B. After signalling ok, the media is also proxied by SER. A Side ------> ------ media goes to SER --<----------media ----------->Media Gateway--------> B side. Possible scenario ; 1-) The BYE is not coming from A side. At that time I see the session on session controller as going on. But infact it is abondened. You are right when the call is on hold. at that time the rtp will not go also. For this session controller must understand when it is on hold.But in any case, I do not know, how will SER end if there is no media session. As I see, the media is controlled by SER. when the bye has come the media session is ended. In fact, When I study on it, I saw that in some cases I saw that, the BYE is coming to soft switch but soft switch send s reply as 500 INTERNAL Error. This is an also issue is related with my soft switch. I have opened a case for this also. Thanks for your interest. I will inform you about the result. Best Regards, ----- Original Message ----- From: Greger V. Teigre To: Hakan YASTI ; serusers(a)lists.iptel.org ; ritesh.j(a)net4.in Sent: Friday, October 28, 2005 6:53 AM Subject: Re: [Serusers] end the session if no media What I'm saying is that you probably want to differ between a long call that after 59 minutes is put on hold and a terminated call where the BYE never came. Both will show no RTP (dependent on UA). Your session controller may be able to tell you if the call is on hold. With regards to No transaction exist: Do you send the BYE directly to the UA and the UA replies with this? If so, I wouldn't worry too much, it just means that you were right in terminating the call because the UA does know of any call. I'm a bit uncertaint about the error though, looks like you may get it from ser (if you send through ser), then the BYE never reaches the UA. (I would expect the UA to respond Call leg doesn't exist). How do you send the BYE that gives this error and to where? g-) ----- Original Message ----- From: Hakan YASTI To: Greger V. Teigre ; serusers(a)lists.iptel.org ; ritesh.j(a)net4.in Sent: Thursday, October 27, 2005 10:25 AM Subject: RE: [Serusers] end the session if no media Yes, I know about it. I can see the active sessions from the session.py. But do not know how to end the seesion, if there is no rtp session and the bye has not come. My session con troller does not check the rtp session because the media is proxied at SER. As I know, may be I am wrong, the ser is managing the media proxy. Ser says to media proxy to start when a call is proxied, and end when the call is finished. Do you suggest a SBC that can do this function..? Thanks, Best Regards, ------------------------------------------------------------------------ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Thursday, October 27, 2005 9:20 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media session.py will show if the session is active. Also when a call is put on hold, there will be no rtp stream. I don't know if your SBC handles that. Or is your question something else? g-) ----- Original Message ----- From: Hakan YASTI To: Greger V. Teigre ; serusers(a)lists.iptel.org ; ritesh.j(a)net4.in Sent: Wednesday, October 26, 2005 09:18 AM Subject: RE: [Serusers] end the session if no media Hi Teigre, Thanks for your reply and kind suggestions. In fact, I am using a session controller for sip calls. And it alerts me via email when a sessiıon is reached for about 60 minutes. At that time, I check it manually and kil the session if the session is abondened. By sending BYE messages to both legs. From the PSTN side I get the reply what I assume ( 200 OK ) but from the sip side I get No transaction exist. Because the session was ended before and session controller couldn’t get the BYE message for some reasons. ( Network failure or something else ). What I am looking for that, from the media proxy, I see the active rtp session when I use media proxy for the calls. And I assume that, when the call has finished and couldn’t send or get BYE message, there will be no rtp stream going on. May be I am wrong for this. If rtp session is still open when the a side hunp up and couldn’t send bye , there is nothing to do except killing the session with sending bye messages to both sides, because there is no way to understand the call is active or not. Best Regards, Hakan ---------------------------------------------------------------------- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, October 26, 2005 10:00 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media Hakan and Ritesh, It seems to me that you are both asking about the same thing. This topic has been discussed many times on serusers. Search google (site:mail.iptel.org) for "BYE stateful sipsak", I'm pretty sure you will get some info. Also, this FAQ could help: http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 g-) ----- Original Message ----- From: Hakan YASTI To: serusers(a)lists.iptel.org Sent: Thursday, October 13, 2005 09:13 AM Subject: [Serusers] end the session if no media Hi, I wonder, is there any way, to end the session when the media has ended. Sometimes, we have problem at BYE message. For some reasons, the bye is not coming and may be it is coming wrong. At that time, the media is ended but the call is seen as active. I am using ser and using media proxy. Any suggestion for this issue ? Best Regards, Hakan YASTI NGN Engineer Teletek Telekomünikasyon Hizmetleri A.Þ. Ayazma Dere Cad. Akþit Plaza No:12 Fulya, Beþiktaþ ISTANBUL - TURKEY Tel : +90 212 310 22 34 Fax : +90 212 227 8700 GSM : +90 533 669 06 84 E-mail: hakan.yasti(a)teletek.net http://www.teletek.net -------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir. Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir. -------------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] subscription
by hgaillac-sip＠yahoo.fr 14 Dec '05

14 Dec '05

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1 0

[Serusers] Help installation of SER!
by carlos moreno 14 Dec '05

14 Dec '05

Good morning, friends, nice to meet you, My name is Carlos Manuel Moreno, from Venezuela, has anyone installed SER, from a RPM file? Which is the correct process? Thanks in advance, Carlos _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 2

Re: [Serusers] Please, help with SUBSCRIBE timeouts
by Andrey Kouprianov 14 Dec '05

14 Dec '05

Yes, that's correct... The UA on that IP address is registered and running... On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > From the trafic it seems, that the authorized SUBSCRIBE is forwarded to > 203.159.32.38. Isn't it correct? > Vaclav > > On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > > > Hi, > > > > Yes, i have tried w/o authorization, but the result is the same. > > The traces are from the client side... As you suggested I dumped the > > traces on the server side with ngrep on both network interface and > > loopback... Loopback yielded no outcome whatsoever (ngrep -W byline -d > > lo0 port 5060). Here are the traces from the from the network > > iterface.... > > > > interface: fxp0 (203.159.31.0/255.255.255.0) > > filter: (ip or ip6) and ( port 5060 ) > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 1 SUBSCRIBE. > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK6789ffc624cbb73dbc918bda9a71956f. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.39:5060 > > SIP/2.0 401 Unauthorized. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 1 SUBSCRIBE. > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.8ccb. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK6789ffc624cbb73dbc918bda9a71956f. > > WWW-Authenticate: Digest realm="sip.interlab.ait.ac.th", > > nonce="439fcec8acac4e38ab93d42a6c610db886870639". > > Server: Sip EXpress router (0.9.3 (i386/freebsd)). > > Content-Length: 0. > > Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=279 > > req_src_ip=203.159.32.39 req_src_port=5060 > > in_uri=sip:12345@sip.interlab.ait.ac.th > > out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1". > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > exit > > 100954 received, 0 dropped > > > > It seems that SER does not even reply with timeout... The timeout is > > probably generated by my JAIN-SIP... Im not sure, anyway... > > > > What is happening here??? > > > > > > > > > > > > On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > >> Hello, > >> have you tried it without authorization? Any changes in behaviour? > >> The network dump is from the machine with SER or from the client? > > (Could > >> you try do such dump on machine with SER on all interfaces including > >> loopback?) > >> Vaclav > >> > >> On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > >> > >>> Hi, > >>> > >>> This is my 3rd email on this issue.. Please, help me with it. > >>> > >>> I am having the problem with SUBSCRIBE requests in ser 0.9.3. I am not > >>> using PA module, > >>> because my PA and Watcher are located on the client side. > >>> > >>> Whenever I try to subscribe to a presence with any entity OTHER than > >>> myself, SER seems to receive the SUBSCRIBE request, however, it does > >>> not forward it. Instead my application keeps on retransmitting > >>> SUBSCRIBEs until SER replies with 408 Timeout... The same things > >>> happen with NOTIFYs. > >>> > >>> I must also mention that it is NOT all the time that SER replies with > >>> timeout... Just most of the times. And only sometimes it lets the > >>> SUBSCRIBE through (for that, I have to restart my app. again and > >>> again, until I get the right result). > >>> > >>> I included ser.cfg + traces. I've tried to use stateless forwarding ( > >>> forward(uri:host, uri:port) ) instead of stateful (as shown in my > >>> ser.cfg), but it didnt work either. > >>> > >>> Please, tell me what is wrong, because I'm out of ideas here... I > >>> would settle even for reinstalling SER, if it's necessary. > >>> > >>> Here's ser.cfg: > >>> > >>> # ----------- global configuration parameters ------------------------ > >>> debug=3 # debug level (cmd line: -dddddddddd) > >>> fork=yes > >>> log_stderror=no # (cmd line: -E) > >>> > >>> /* Uncomment these lines to enter debugging mode > >>> fork=no > >>> log_stderror=yes > >>> */ > >>> > >>> check_via=no # (cmd. line: -v) > >>> dns=no # (cmd. line: -r) > >>> rev_dns=no # (cmd. line: -R) > >>> #port=5060 > >>> #children=4 > >>> fifo="/tmp/ser_fifo" > >>> fifo_mode=0766 > >>> > >>> sip_warning=yes > >>> server_signature=yes > >>> > >>> alias=sip.interlab.ait.ac.th > >>> > >>> # ------------------ module loading ---------------------------------- > >>> # Uncomment this if you want to use SQL database > >>> loadmodule "/usr/local/lib/ser/modules/mysql.so" > >>> loadmodule "/usr/local/lib/ser/modules/sl.so" > >>> loadmodule "/usr/local/lib/ser/modules/tm.so" > >>> loadmodule "/usr/local/lib/ser/modules/rr.so" > >>> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > >>> loadmodule "/usr/local/lib/ser/modules/usrloc.so" > >>> loadmodule "/usr/local/lib/ser/modules/registrar.so" > >>> loadmodule "/usr/local/lib/ser/modules/textops.so" > >>> loadmodule "/usr/local/lib/ser/modules/auth.so" > >>> loadmodule "/usr/local/lib/ser/modules/auth_db.so" > >>> loadmodule "/usr/local/lib/ser/modules/uri.so" > >>> loadmodule "/usr/local/lib/ser/modules/uri_db.so" > >>> loadmodule "/usr/local/lib/ser/modules/exec.so" > >>> loadmodule "/usr/local/lib/ser/modules/pa.so" > >>> > >>> # ----------------- setting module-specific parameters --------------- > >>> modparam("usrloc", "db_mode", 2) > >>> modparam("auth_db", "calculate_ha1", yes) > >>> modparam("auth_db", "password_column", "password") > >>> modparam("rr", "enable_full_lr", 1) > >>> > >>> # ------------------------- request routing logic ------------------- > >>> route{ > >>> > >>> # initial sanity checks -- messages with > >>> # max_forwards==0, or excessively long requests > >>> if (!mf_process_maxfwd_header("10")) { > >>> log(1, "Message has too many hops. Message disgarted"); > >>> sl_send_reply("483","Too Many Hops"); > >>> break; > >>> }; > >>> if (msg:len >= 4096 ) { > >>> log(1, "Message too big. Message disgarted"); > >>> sl_send_reply("513", "Message too big"); > >>> break; > >>> }; > >>> > >>> #if (method!="REGISTER" ) record_route(); > >>> > >>> #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 (uri==myself) { > >>> if (method=="REGISTER") { > >>> if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { > >>> log(1, "Authentication for REGISTER"); > >>> www_challenge("sip.interlab.ait.ac.th", "0"); > >>> break; > >>> }; > >>> > >>> #if(!is_user("replicator") & !check_to() ){ > >>> # log(1, "Unregistered user registration attempt\n"); > >>> # sl_send_reply("403", "Forbidden - Registered > >>> users only"); > >>> # break; > >>> #}; > >>> > >>> if(!save("location")) { > >>> log(1, "Error saving AOR"); > >>> sl_reply_error(); > >>> }; > >>> break; > >>> }; > >>> > >>> lookup("aliases"); > >>> if (uri!=myself) { > >>> append_hf("P-hint: outbound alias\r\n"); > >>> route(1); > >>> break; > >>> }; > >>> > >>> #Handle local offline or non-existent users > >>> if(method=="INVITE") { > >>> if(!lookup("location") ) { > >>> log(1, "INVITE for an offline user processing"); > >>> route(3); #offline and non-existent users > >>> break; > >>> }; > >>> }; > >>> > >>> if(method=="SUBSCRIBE") { > >>> if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { > >>> log(1, "Authentication for SUBSCRIBE"); > >>> www_challenge("sip.interlab.ait.ac.th", "0"); > >>> break; > >>> }; > >>> > >>> if(!lookup("location") ) { > >>> log(1, "No registered user found for SUBSCRIBE"); > >>> > >>> sl_send_reply("404", "Not Found"); > >>> break; > >>> }; > >>> }; > >>> > >>> # native SIP destinations are handled using our USRLOC DB > >>> if(method=="NOTIFY") { > >>> if (!lookup("location")) { > >>> sl_send_reply("404", "Not Found"); > >>> break; > >>> }; > >>> > >>> log(1, "NOTIFY reveived"); > >>> }; > >>> }; > >>> > >>> 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(); > >>> }; > >>> } > >>> > >>> # ------------ Process offline and non-existent users ---------- > >>> route[3] { > >>> # do not continue if it is a retransmission > >>> if(!t_newtran()) { > >>> sl_reply_error(); > >>> break; > >>> }; > >>> > >>> # use external script to send an email notification in case > >>> # user is offline, or just reply back with 404 (Not Found) if > >>> # the user is not a registered user > >>> if(!exec_msg('/home/andrey/sip_email.sh') ) { > >>> t_reply("404", "Not Found - User does not exist"); > >>> } > >>> else { > >>> t_reply("404", "Not Found - User offline"); > >>> }; > >>> break; > >>> } > >>> > >>> . . . . . . . . > >>> > >>> > >>> Here are the traces: > >>> > >>> Message 1 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 1 SUBSCRIBE > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 2 from 203.159.31.36:5060 to 203.159.32.39:5060 > >>> > >>> > >>> SIP/2.0 401 Unauthorized > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 1 SUBSCRIBE > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.5e75 > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 > >>> WWW-Authenticate: Digest > >>> realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0" > >>> Server: Sip EXpress router (0.9.3 (i386/freebsd)) > >>> Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=17754 > >>> req_src_ip=203.159.32.39 req_src_port=5060 > >>> in_uri=sip:12345@sip.interlab.ait.ac.th > >>> out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1" > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 3 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 4 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 5 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> . . . . . . > >>> > >>> And so it goes, until SER replies with Timeout. > >>> > >>> Can you tell me what is wrong??? > >>> > >>> Andrey. > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >>> > >> > > >

2 11

[Serusers] SER with presence release
by Andrey Kouprianov 14 Dec '05

14 Dec '05

Hi, Dont mean to bother much, but when is stable SER with presence support will be made available?

3 5

[Serusers] SUBSCRIBEs and NOTIFYs timeout
by Andrey Kouprianov 14 Dec '05

14 Dec '05

Hi, Im using ser 0.9.3. The PA and Watcher are located on my UA, so basically my SUBSCRIBEs and NOTIFYs just need to be forwarded between clients. Here's a part of ser.cfg that does the SUBSCRIBE and NOTIFY processing (please, note that I disabled the authentication for now): . . . . . . if(method=="SUBSCRIBE") { if(!lookup("location") ) { log(1, "No registered user found for SUBSCRIBE request"); sl_send_reply("404", "Not Found"); break; }; }; if(method=="NOTIFY") { if (!lookup("location")) { log(1, "No registered user found for NOTIFY request"); sl_send_reply("404", "Not Found"); break; }; }; if( !t_relay() ) { sl_reply_error(); }; . . . . . . . . The problem is that only SOMETIMES SUIBSCRIBEs and NOTIFYs get through without any problem. However, MOST OF THE TIMES they get stuck in the server. The message log does not show any errors and tracesviewer tool shows only retransmissions for the issued requests. In the end SER replies with 408 Timeout response. I have tried to use forward(uri:host, uri:port) instead of t_relay(), but result is the same. I have used eyeBeam to do the presence subscriptions, and the result is the same as with my UA... Can you, please, explain to me what might have happened??? I am out of ideas. Andrey.

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Re: [Serusers] Please, help with SUBSCRIBE timeouts
by Andrey Kouprianov 14 Dec '05

14 Dec '05

So, what could be the problem then??? On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > > It seems that SER does not even reply with timeout... The timeout is > > probably generated by my JAIN-SIP... Im not sure, anyway... > > > > What is happening here??? > > > > Yes, you are right. SER forwards the message but the other side doesn't > respond and the client itself generates a timeout (I think, that timeouts > should be generated by SER too, but may be, that the client has lower > timer values). > Vaclav > > P.S. I missed this part of your message... :-) > > > > > > > > > > > On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > >> Hello, > >> have you tried it without authorization? Any changes in behaviour? > >> The network dump is from the machine with SER or from the client? > > (Could > >> you try do such dump on machine with SER on all interfaces including > >> loopback?) > >> Vaclav > >> > >> On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > >> > >>> Hi, > >>> > >>> This is my 3rd email on this issue.. Please, help me with it. > >>> > >>> I am having the problem with SUBSCRIBE requests in ser 0.9.3. I am not > >>> using PA module, > >>> because my PA and Watcher are located on the client side. > >>> > >>> Whenever I try to subscribe to a presence with any entity OTHER than > >>> myself, SER seems to receive the SUBSCRIBE request, however, it does > >>> not forward it. Instead my application keeps on retransmitting > >>> SUBSCRIBEs until SER replies with 408 Timeout... The same things > >>> happen with NOTIFYs. > >>> > >>> I must also mention that it is NOT all the time that SER replies with > >>> timeout... Just most of the times. And only sometimes it lets the > >>> SUBSCRIBE through (for that, I have to restart my app. again and > >>> again, until I get the right result). > >>> > >>> I included ser.cfg + traces. I've tried to use stateless forwarding ( > >>> forward(uri:host, uri:port) ) instead of stateful (as shown in my > >>> ser.cfg), but it didnt work either. > >>> > >>> Please, tell me what is wrong, because I'm out of ideas here... I > >>> would settle even for reinstalling SER, if it's necessary. > >>> > >>> Here's ser.cfg: > >>> > >>> # ----------- global configuration parameters ------------------------ > >>> debug=3 # debug level (cmd line: -dddddddddd) > >>> fork=yes > >>> log_stderror=no # (cmd line: -E) > >>> > >>> /* Uncomment these lines to enter debugging mode > >>> fork=no > >>> log_stderror=yes > >>> */ > >>> > >>> check_via=no # (cmd. line: -v) > >>> dns=no # (cmd. line: -r) > >>> rev_dns=no # (cmd. line: -R) > >>> #port=5060 > >>> #children=4 > >>> fifo="/tmp/ser_fifo" > >>> fifo_mode=0766 > >>> > >>> sip_warning=yes > >>> server_signature=yes > >>> > >>> alias=sip.interlab.ait.ac.th > >>> > >>> # ------------------ module loading ---------------------------------- > >>> # Uncomment this if you want to use SQL database > >>> loadmodule "/usr/local/lib/ser/modules/mysql.so" > >>> loadmodule "/usr/local/lib/ser/modules/sl.so" > >>> loadmodule "/usr/local/lib/ser/modules/tm.so" > >>> loadmodule "/usr/local/lib/ser/modules/rr.so" > >>> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > >>> loadmodule "/usr/local/lib/ser/modules/usrloc.so" > >>> loadmodule "/usr/local/lib/ser/modules/registrar.so" > >>> loadmodule "/usr/local/lib/ser/modules/textops.so" > >>> loadmodule "/usr/local/lib/ser/modules/auth.so" > >>> loadmodule "/usr/local/lib/ser/modules/auth_db.so" > >>> loadmodule "/usr/local/lib/ser/modules/uri.so" > >>> loadmodule "/usr/local/lib/ser/modules/uri_db.so" > >>> loadmodule "/usr/local/lib/ser/modules/exec.so" > >>> loadmodule "/usr/local/lib/ser/modules/pa.so" > >>> > >>> # ----------------- setting module-specific parameters --------------- > >>> modparam("usrloc", "db_mode", 2) > >>> modparam("auth_db", "calculate_ha1", yes) > >>> modparam("auth_db", "password_column", "password") > >>> modparam("rr", "enable_full_lr", 1) > >>> > >>> # ------------------------- request routing logic ------------------- > >>> route{ > >>> > >>> # initial sanity checks -- messages with > >>> # max_forwards==0, or excessively long requests > >>> if (!mf_process_maxfwd_header("10")) { > >>> log(1, "Message has too many hops. Message disgarted"); > >>> sl_send_reply("483","Too Many Hops"); > >>> break; > >>> }; > >>> if (msg:len >= 4096 ) { > >>> log(1, "Message too big. Message disgarted"); > >>> sl_send_reply("513", "Message too big"); > >>> break; > >>> }; > >>> > >>> #if (method!="REGISTER" ) record_route(); > >>> > >>> #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 (uri==myself) { > >>> if (method=="REGISTER") { > >>> if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { > >>> log(1, "Authentication for REGISTER"); > >>> www_challenge("sip.interlab.ait.ac.th", "0"); > >>> break; > >>> }; > >>> > >>> #if(!is_user("replicator") & !check_to() ){ > >>> # log(1, "Unregistered user registration attempt\n"); > >>> # sl_send_reply("403", "Forbidden - Registered > >>> users only"); > >>> # break; > >>> #}; > >>> > >>> if(!save("location")) { > >>> log(1, "Error saving AOR"); > >>> sl_reply_error(); > >>> }; > >>> break; > >>> }; > >>> > >>> lookup("aliases"); > >>> if (uri!=myself) { > >>> append_hf("P-hint: outbound alias\r\n"); > >>> route(1); > >>> break; > >>> }; > >>> > >>> #Handle local offline or non-existent users > >>> if(method=="INVITE") { > >>> if(!lookup("location") ) { > >>> log(1, "INVITE for an offline user processing"); > >>> route(3); #offline and non-existent users > >>> break; > >>> }; > >>> }; > >>> > >>> if(method=="SUBSCRIBE") { > >>> if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { > >>> log(1, "Authentication for SUBSCRIBE"); > >>> www_challenge("sip.interlab.ait.ac.th", "0"); > >>> break; > >>> }; > >>> > >>> if(!lookup("location") ) { > >>> log(1, "No registered user found for SUBSCRIBE"); > >>> > >>> sl_send_reply("404", "Not Found"); > >>> break; > >>> }; > >>> }; > >>> > >>> # native SIP destinations are handled using our USRLOC DB > >>> if(method=="NOTIFY") { > >>> if (!lookup("location")) { > >>> sl_send_reply("404", "Not Found"); > >>> break; > >>> }; > >>> > >>> log(1, "NOTIFY reveived"); > >>> }; > >>> }; > >>> > >>> 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(); > >>> }; > >>> } > >>> > >>> # ------------ Process offline and non-existent users ---------- > >>> route[3] { > >>> # do not continue if it is a retransmission > >>> if(!t_newtran()) { > >>> sl_reply_error(); > >>> break; > >>> }; > >>> > >>> # use external script to send an email notification in case > >>> # user is offline, or just reply back with 404 (Not Found) if > >>> # the user is not a registered user > >>> if(!exec_msg('/home/andrey/sip_email.sh') ) { > >>> t_reply("404", "Not Found - User does not exist"); > >>> } > >>> else { > >>> t_reply("404", "Not Found - User offline"); > >>> }; > >>> break; > >>> } > >>> > >>> . . . . . . . . > >>> > >>> > >>> Here are the traces: > >>> > >>> Message 1 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 1 SUBSCRIBE > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 2 from 203.159.31.36:5060 to 203.159.32.39:5060 > >>> > >>> > >>> SIP/2.0 401 Unauthorized > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 1 SUBSCRIBE > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.5e75 > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 > >>> WWW-Authenticate: Digest > >>> realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0" > >>> Server: Sip EXpress router (0.9.3 (i386/freebsd)) > >>> Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=17754 > >>> req_src_ip=203.159.32.39 req_src_port=5060 > >>> in_uri=sip:12345@sip.interlab.ait.ac.th > >>> out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1" > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 3 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 4 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 5 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> . . . . . . > >>> > >>> And so it goes, until SER replies with Timeout. > >>> > >>> Can you tell me what is wrong??? > >>> > >>> Andrey. > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >>> > >> > > >

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[Serusers] Please, help with SUBSCRIBE timeouts
by Andrey Kouprianov 14 Dec '05

14 Dec '05

Hi, This is my 3rd email on this issue.. Please, help me with it. I am having the problem with SUBSCRIBE requests in ser 0.9.3. I am not using PA module, because my PA and Watcher are located on the client side. Whenever I try to subscribe to a presence with any entity OTHER than myself, SER seems to receive the SUBSCRIBE request, however, it does not forward it. Instead my application keeps on retransmitting SUBSCRIBEs until SER replies with 408 Timeout... The same things happen with NOTIFYs. I must also mention that it is NOT all the time that SER replies with timeout... Just most of the times. And only sometimes it lets the SUBSCRIBE through (for that, I have to restart my app. again and again, until I get the right result). I included ser.cfg + traces. I've tried to use stateless forwarding ( forward(uri:host, uri:port) ) instead of stateful (as shown in my ser.cfg), but it didnt work either. Please, tell me what is wrong, because I'm out of ideas here... I would settle even for reinstalling SER, if it's necessary. Here's ser.cfg: # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_mode=0766 sip_warning=yes server_signature=yes alias=sip.interlab.ait.ac.th # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/pa.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { log(1, "Message has too many hops. Message disgarted"); sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 4096 ) { log(1, "Message too big. Message disgarted"); sl_send_reply("513", "Message too big"); break; }; #if (method!="REGISTER" ) record_route(); #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 (uri==myself) { if (method=="REGISTER") { if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { log(1, "Authentication for REGISTER"); www_challenge("sip.interlab.ait.ac.th", "0"); break; }; #if(!is_user("replicator") & !check_to() ){ # log(1, "Unregistered user registration attempt\n"); # sl_send_reply("403", "Forbidden - Registered users only"); # break; #}; if(!save("location")) { log(1, "Error saving AOR"); sl_reply_error(); }; break; }; lookup("aliases"); if (uri!=myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; #Handle local offline or non-existent users if(method=="INVITE") { if(!lookup("location") ) { log(1, "INVITE for an offline user processing"); route(3); #offline and non-existent users break; }; }; if(method=="SUBSCRIBE") { if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { log(1, "Authentication for SUBSCRIBE"); www_challenge("sip.interlab.ait.ac.th", "0"); break; }; if(!lookup("location") ) { log(1, "No registered user found for SUBSCRIBE"); sl_send_reply("404", "Not Found"); break; }; }; # native SIP destinations are handled using our USRLOC DB if(method=="NOTIFY") { if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; log(1, "NOTIFY reveived"); }; }; 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(); }; } # ------------ Process offline and non-existent users ---------- route[3] { # do not continue if it is a retransmission if(!t_newtran()) { sl_reply_error(); break; }; # use external script to send an email notification in case # user is offline, or just reply back with 404 (Not Found) if # the user is not a registered user if(!exec_msg('/home/andrey/sip_email.sh') ) { t_reply("404", "Not Found - User does not exist"); } else { t_reply("404", "Not Found - User offline"); }; break; } . . . . . . . . Here are the traces: Message 1 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 1 SUBSCRIBE From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Content-Length: 0 Message 2 from 203.159.31.36:5060 to 203.159.32.39:5060 SIP/2.0 401 Unauthorized Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 1 SUBSCRIBE From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.5e75 Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 WWW-Authenticate: Digest realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0" Server: Sip EXpress router (0.9.3 (i386/freebsd)) Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=17754 req_src_ip=203.159.32.39 req_src_port=5060 in_uri=sip:12345@sip.interlab.ait.ac.th out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1" Content-Length: 0 Message 3 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 Message 4 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 Message 5 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 . . . . . . And so it goes, until SER replies with Timeout. Can you tell me what is wrong??? Andrey.

2 2

[Users] inbound call failed
by unplug 14 Dec '05

14 Dec '05

Below is the common configuration of the network. Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger) I can make call from softphone to Telephone. However, it is failed to make call from Telephone to softphone. I wonder why it happened and any reference to trace the problem. Anyone have such experience?

2 4

[Serusers] Using SER together with Asterisk
by bharat.sarvan＠wipro.com 14 Dec '05

14 Dec '05

Hello all, I am trying to integrate SER and Asterisk. So could anybody please guide me with the changes and the configuration that I need to in either SER or Asterisk for achieving it. Like for suppose I want to achieve a call being placed from a User 1 registered to SER to User 2 who is registered to Asterisk.. So how do I move ahead to achieve the same. And are there any other new things that one can come up with using SER and Asterisk together which can't be achieved using SER or Asterisk alone? Kindly do let me know. Thanks and Regards, Bharat Sarvan The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

Re: [Serusers] simple click-to-call
by Kapil Dhawan 14 Dec '05

14 Dec '05

If SER is running, that means fifo is ON. >From: Ken Rozinsky <xraycharlie(a)gmail.com> >To: Kapil Dhawan <sersavvy(a)hotmail.com> >Subject: Re: [Serusers] simple click-to-call >Date: Tue, 13 Dec 2005 11:24:58 -0700 > >Thanks for the info. > >A question for you if you have the time: > >The script checks if it can write to /tmp/ser_fifo and fails. >The comment mention having to have the fifo server running. How do I >check if this is running or not? > >I see in the documantation a mention of fifo_db but I don;t think I >need this as am not worried about persistance. > >The permissions on /tmp/ser_fifo are currently prw-rw---- > >I tried running the script as root and get the write error also. > >Thanks again. > >Regards > >Ken > > >On 12/12/05, Kapil Dhawan <sersavvy(a)hotmail.com> wrote: > > Ken > > > > Click-to-dial is present in example directoy as ctd.sh. It takes two > > arguments From and To. To run it from Web-interface, you need to run >this > > command thru script. > > > > Regards > > > > > > >From: Ken Rozinsky <xraycharlie(a)gmail.com> > > >To: serusers(a)lists.iptel.org > > >Subject: [Serusers] simple click-to-call > > >Date: Mon, 12 Dec 2005 17:27:18 -0700 > > > > > >Hello, > > > > > >As an exercise in learning how all this stuff works, I'm trying to set > > >up a simple click-to-call service for our office to use. And > > >potentialy to allow clients to use vai the web site to save them on > > >the long distance charges. > > > > > >I set up Asterisk with the TACI script and got it to all work with an > > >account on gafachi but I noticed that it could only handle one call at > > >a time. > > > > > >I was told that openSer would allow more than one call at a time so > > >I've installed it (basic no mySql) and have it up and running. > > > > > >Now, being new to this I have no idea how to connct to gafachi and do > > >a click-to-call type thing. > > > > > >Would anybody be able to help me with this? > > > > > >Regards > > > > > >_______________________________________________ > > >Serusers mailing list > > >serusers(a)lists.iptel.org > > >http://lists.iptel.org/mailman/listinfo/serusers > > > > _________________________________________________________________ > > 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

1 0

Re: [Serusers] using pa module
by lo ™ 14 Dec '05

14 Dec '05

ok, i set up eyebeam with peer-to-peer mode in the presence agent settings but i don't understand yet if it is close to work or not... thank you, again! this is my log: Maxfwd module- initializing db_url=0x8208c78 db_url=mysql://ser:heslo@localhost/ser db_url.len=31 pa_domain=<domain.tld> pa_mod: use_db=1 db_url.s=mysql://ser:heslo@localhost/ser pa_domain=< domain.tld> pa_mod_init done subscribe_fixup: pdomain name is registrar subscribe_fixup: pdomain name is registrar INFO: udp_init: SO_RCVBUF is initially 108544 INFO: udp_init: SO_RCVBUF is finally 217088 INFO: udp_init: SO_RCVBUF is initially 108544 INFO: udp_init: SO_RCVBUF is finally 217088 INFO: fifo process starting: 28887 SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... handle_subscription() entered get_pres_uri: _puri=sip:ennevu@<domain.tld> handle_subscription() entered get_pres_uri: _puri=sip:lo2@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=11@<domain.tld> presid=1 new_presentity_no_wb=0xb618bdb8 for uri=11@<domain.tld> add_presentity _p=0xb618bdb8 p_uri=11@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=ennevu@< domain.tld> presid=2 new_presentity_no_wb=0xb618be08 for uri=ennevu@<domain.tld> add_presentity _p=0xb618be08 p_uri=ennevu@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=lo2@<domain.tld> presid=3 new_presentity_no_wb=0xb618be60 for uri=lo2@<domain.tld> add_presentity _p=0xb618be60 p_uri=lo2@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=10@<domain.tld> presid=4 new_presentity_no_wb=0xb618beb0 for uri=10@<domain.tld> add_presentity _p=0xb618beb0 p_uri=10@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=lo@<domain.tld> presid=5 new_presentity_no_wb=0xb618bf00 for uri=lo@<domain.tld> add_presentity _p=0xb618bf00 p_uri=lo@<domain.tld> db_read_watcherinfo: _p->uri='11@<domain.tld>' db_read_watcherinfo: _p->uri='11@<domain.tld>' done db_read_watcherinfo: _p->uri='ennevu@<domain.tld>' db_read_watcherinfo: _p->uri='ennevu@<domain.tld>' done db_read_watcherinfo: _p->uri='lo2@<domain.tld>' db_read_watcherinfo: _p->uri='lo2@<domain.tld>' done db_read_watcherinfo: _p->uri='10@<domain.tld>' db_read_watcherinfo: _p->uri='10@<domain.tld>' done db_read_watcherinfo: _p->uri='lo@<domain.tld>' db_read_watcherinfo: _p->uri='lo@<domain.tld>' done new_watcher starting new_watcher: _p->uri=ennevu@<domain.tld> db_new_watcher: watcher->uri=11@<domain.tld> new_watcher: watcher->package=presence new_watcher: getting values: res=0x81019b8 res->n=1 new_watcher: status=active handle_subscription about to return 1: w->event_package=1 w->accept=2 p->flags=4 w->flags=1 w=0xb618c490 new_watcher starting new_watcher: _p->uri=lo2@<domain.tld> db_new_watcher: watcher->uri=11@<domain.tld> new_watcher: watcher->package=presence new_watcher: getting values: res=0x8101998 res->n=1 new_watcher: status=active handle_subscription about to return 1: w->event_package=1 w->accept=2 p->flags=4 w->flags=1 w=0xb618c928 handle_subscription() entered get_pres_uri: _puri=sip:lo@<domain.tld> new_watcher starting new_watcher: _p->uri=lo@<domain.tld> db_new_watcher: watcher->uri=10@<domain.tld> new_watcher: watcher->package=presence new_watcher: getting values: res=0x810a640 res->n=1 new_watcher: status=active handle_subscription about to return 1: w->event_package=1 w->accept=2 p->flags=4 w->flags=1 w=0xb618fb38 handle_subscription() entered get_pres_uri: _puri=sip:11@<domain.tld> new_watcher starting new_watcher: _p->uri=11@<domain.tld> db_new_watcher: watcher->uri=10@<domain.tld> new_watcher: watcher->package=presence new_watcher: getting values: res=0x810a500 res->n=1 new_watcher: status=active handle_subscription about to return 1: w->event_package=1 w->accept=2 p->flags=4 w->flags=1 w=0xb6191980 timer_presentity: _p=0xb618bdb8 11@<domain.tld> flags=4 watchers=0xb6191980 w=0xb6191980 10@<domain.tld>ennevu<sip:10@<domain.tld>> flags=1 notifying 10@<domain.tld> _p->flags=4 _w->event_package=1 _w->accept=2 _w->status=1 send_pidf_notify timer_presentity: _p=0xb618be08 ennevu@<domain.tld> flags=4 watchers=0xb618c490 w=0xb618c490 11@<domain.tld>lo<sip:11@<domain.tld>> flags=1 notifying 11@<domain.tld> _p->flags=4 _w->event_package=1 _w->accept=2 _w->status=1 send_pidf_notify timer_presentity: _p=0xb618be60 lo2@<domain.tld> flags=4 watchers=0xb618c928 w=0xb618c928 11@<domain.tld>lo<sip:11@<domain.tld>> flags=1 notifying 11@<domain.tld> _p->flags=4 _w->event_package=1 _w->accept=2 _w->status=1 send_pidf_notify timer_presentity: _p=0xb618bf00 lo@<domain.tld> flags=4 watchers=0xb618fb38 w=0xb618fb38 10@<domain.tld>ennevu<sip:10@<domain.tld>> flags=1 notifying 10@<domain.tld> _p->flags=4 _w->event_package=1 _w->accept=2 _w->status=1 send_pidf_notify Warning: run_failure_handlers: no UAC support (0, 0) Warning: run_failure_handlers: no UAC support (0, 0) Warning: run_failure_handlers: no UAC support (0, 0) Warning: run_failure_handlers: no UAC support (0, 0) Removing watcher 10@<domain.tld> notifying 10@<domain.tld> _p->flags=0 _w->event_package=1 _w->accept=2 _w->status=1 Removing watcher 11@<domain.tld> notifying 11@<domain.tld> _p->flags=0 _w->event_package=1 _w->accept=2 _w->status=1 Removing watcher 11@<domain.tld> notifying 11@<domain.tld> _p->flags=0 _w->event_package=1 _w->accept=2 _w->status=1 Removing watcher 10@<domain.tld> notifying 10@<domain.tld> _p->flags=0 _w->event_package=1 _w->accept=2 _w->status=1

2 1

[Serusers] Mediaproxy Call hold/unhold
by tcchan 14 Dec '05

14 Dec '05

Dear All, I encountered a problem like this. Case 1: ------- If UA1 calls UA2 using the following: Normal call: UA1 ------invite -------port 30000 -----> UA2 UA1 <--------OK---port 26000--------------UA2 The connection and media is established. Hold the call: UA1------invite ---port 30000 (hold)--> UA2 UA1 <----------OK port 26000----------UA2 The media is in hold state. Unhold the call: UA1------invite ----port 30000--------->UA2 UA1 <---------OK port 260000----------UA2 Media resume. The above mentioned work fine with mediaproxy. Case 2: ------- If UA1 calls UA2 using the following: Normal call: UA1 ------invite -------port 30000 -----> UA2 UA1 <--------OK---port 26000--------------UA2 The connection and media is established. Hold the call: UA1------invite ---port 30000 (hold)--> UA2 UA1 <----------OK port 26000----------UA2 The media is in hold state. Unhold the call: UA1------invite ----port 30001--------->UA2 UA1 <---------OK port 260000----------UA2 *****Noticed that the reinvite has a different port! In this situation, media does not resume and mediaproxy has a "stray hold" session. I tested case2 without flowing through the mediaproxy and found that the call can be hold/unhold successfully. I am not familiar with RFC at all, please help! (1) I would like to know whether it is valid to reinvite with a different port as in case 2, i.e., is the UA1 ill-behave in case 2 when unhold the call with different port? (2) Is the mediaproxy designed to work using the some port when hold/unhold. Thanks you very much for any help available. Regards, TC Chan

1 0

[Users] failure_route.... I could cry...
by Douglas Garstang 14 Dec '05

14 Dec '05

I am trying to have OpenSER forward registration requests to multiple Asterisk boxes with failover. Each Asterisk box has two ip addresses. My logic tries to relay to the first IP on the first Asterisk box (192.168.10.100). If that fails, it tries to relay to the second IP on the first Asterisk box (192.168.10.17). It then goes on and repeats the process with another Asterisk box who's IP's are 192.168.10.10.101 and 192.168.10.8. Actually the first IP address on each Asterisk system is fake (100 and 101)... that's how I am testing that it will fail to the second one. The openser.cfg below is causing this to be logged to messages: Dec 13 17:50:08 bil-pdev-3 openser[22318]: route10: Asterisk-1, NIC-1 Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: Asterisk-2, NIC-1 Dec 13 17:50:08 bil-pdev-3 openser[22318]: ERROR: t_newtran: transaction already in process 0xb6126560 < --- Why? Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: t_relay returned error <--- Why? Dec 13 17:50:09 bil-pdev-3 openser[22328]: failure21: Failed to register with Asterisk-2, NIC-1 Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: Asterisk-2, NIC-2 Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: End of routine Dec 13 17:50:10 bil-pdev-3 openser[22322]: User local found Dec 13 17:50:10 bil-pdev-3 openser[22320]: User local found Why am I getting the t_newtran and t_relay errors in there? What am I doing wrong? I'm just calling rewritehostport(next-ip), followed by append_branch() and t_relay. In fact something really weird happens. When I have the code there for route[20], failure_route[21] and route[22], it causes the code for the first three routing blocks, route[10], failure_route[11] and route[12] to act differently. It seems to make route20 fail to connect eventhough it comes AFTER it in the code. I'd really appreciate some help! Am I doing something wrong with rewritehostport and append_branch, because quite frankly the docs are terrible. if ( method == REGISTER ) { if ( !www_authorize("voip.com", "subscriber") ) { www_challenge("voip.com", "0"); }; save("location"); route(10); route(20); # # First Asterisk System. # # Try first IP on this system (fake - will fail) route[10] { xlog ("L_INFO","route10: Asterisk-1, NIC-1"); t_on_failure("11"); rewritehostport("192.168.10.100:5060"); append_branch(); if ( !t_relay() ) { xlog ("L_INFO","route10: t_relay returned error"); } } failure_route[11] { xlog ("L_INFO","failure11: Failed to register with Asterisk-1, NIC-1"); route(12); } # Try second IP on this system (real - should work) route[12] { xlog ("L_INFO","route12: Asterisk-1, NIC-2"); t_on_failure("13"); rewritehostport("192.168.10.17:5060"); append_branch(); if ( !t_relay() ) { xlog ("L_INFO","route12: t_relay returned error"); } xlog ("L_INFO","route12: End of routine"); return; } # # Second Asterisk System # # Try first IP on this system (fake - will fail) route[20] { xlog ("L_INFO","route20: Asterisk-2, NIC-1"); t_on_failure("21"); rewritehostport("192.168.10.101:5060"); append_branch(); if ( !t_relay() ) { xlog ("L_INFO","route20: t_relay returned error"); } } failure_route[21] { xlog ("L_INFO","failure21: Failed to register with Asterisk-2, NIC-1"); route(22); } # Try second IP on this system (real - should work) route[22] { xlog ("L_INFO","route22: Asterisk-2, NIC-2"); t_on_failure("23"); rewritehostport("192.168.10.8:5060"); append_branch(); if ( !t_relay() ) { xlog ("L_INFO","route22: t_relay returned error"); } xlog ("L_INFO","route22: End of routine"); return; } Doug

1 0

[Serusers] simple click-to-call
by Ken Rozinsky 14 Dec '05

14 Dec '05

Hello, As an exercise in learning how all this stuff works, I'm trying to set up a simple click-to-call service for our office to use. And potentialy to allow clients to use vai the web site to save them on the long distance charges. I set up Asterisk with the TACI script and got it to all work with an account on gafachi but I noticed that it could only handle one call at a time. I was told that openSer would allow more than one call at a time so I've installed it (basic no mySql) and have it up and running. Now, being new to this I have no idea how to connct to gafachi and do a click-to-call type thing. Would anybody be able to help me with this? Regards

2 2

[Serusers] Changing Sip Port
by sagar 13 Dec '05

13 Dec '05

Hi All, I am using ser 0.8.14 +asterisk B2bua. Currently softphone is registering on standard sip port 5060 . Is it possibe to use any other port like 5090 or 8069 I tried to do so by changing the listining port to 5090 in ser.cfg but in case the softphone is not getting register. Here is the ser log 0(11348) SIP Request: 0(11348) method: <REGISTER> 0(11348) uri: <sip:192.168.1.100> 0(11348) version: <SIP/2.0> 0(11348) parse_headers: flags=1 0(11348) end of header reached, state=5 0(11348) parse_headers: Via found, flags=1 0(11348) parse_headers: this is the first via 0(11348) After parse_msg... 0(11348) preparing to run routing scripts... 0(11348) DEBUG : is_maxfwd_present: searching for max_forwards header 0(11348) parse_headers: flags=128 0(11348) DEBUG: is_maxfwd_present: value = 70 0(11348) parse_headers: flags=8 0(11348) DEBUG: add_param: tag=fa37b062306344d68023aed26289e820 0(11348) DEBUG: add_param: epid=f657995ecc 0(11348) end of header reached, state=29 0(11348) parse_headers: flags=256 0(11348) end of header reached, state=9 0(11348) DEBUG: get_hdr_field: <To> [37]; uri=[sip:2851816416@192.168.1.100:5090] 0(11348) DEBUG: to body [<sip:2851816416@192.168.1.100:5090> ] 0(11348) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 0(11348) DEBUG: get_hdr_body : content_length=0 0(11348) found end of header 0(11348) find_first_route(): No Route headers found 0(11348) loose_route(): There is no Route HF 0(11348) parse_headers: flags=64 0(11348) parse_headers: flags=64 0(11348) parse_headers: flags=33554432 0(11348) check_self - checking if host==us: 13==13 && [192.168.1.100] == [192.168.1.100] 0(11348) check_self - checking if port 5090 matches port 5060 0(11348) check_self: host != me 0(11348) DEBUG: t_addifnew: msg id=1 , global msg id=0 , T on entrance=0xffffffff 0(11348) parse_headers: flags=-1 0(11348) parse_headers: flags=60 0(11348) t_lookup_request: start searching: hash=58712, isACK=0 0(11348) DEBUG: proceeding to pre-RFC3261 transaction matching 0(11348) DEBUG: t_lookup_request: no transaction found 0(11348) DEBUG: mk_proxy: doing DNS lookup... 0(11348) check_via_address(192.168.1.2, 192.168.1.2, 0) 0(11348) DEBUG: add_to_tail_of_timer[4]: 0xb6165fe8 0(11348) DEBUG: add_to_tail_of_timer[0]: 0xb6165ffc 0(11348) SER: new transaction fwd'ed 0(11348) DEBUG:destroy_avp_list: destroing list (nil) 0(11348) receive_msg: cleaning up Please advise. Thanks In Advance. --------------------------------- Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Yahoo! Personals

4 3

[Serusers] Segmentation Fault (SER Not Starting)
by sagar 13 Dec '05

13 Dec '05

Hi All, I am having trouble when trying to start SER. I am using Ser 0.8.14 on Redhat ES 3. When I am trying to start it gives Segmentation Fault. Can anybody suggess how to overcome this problem. Thanks And Regards Vidhya Sagar Dixit --------------------------------- Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping

4 3

[Serusers] Getting Nokia 6680 PTT to work
by Pieter De Wit 13 Dec '05

13 Dec '05

Hello Guys, Please can someone give me some pointers as to how I can get this to work - I have looked at the guide and it's a bit big for a "browse" - Isn't there a mini-how-to ? I need the server to access reg's etc. Thanks, Pieter �This e-mail is sent on the Terms and Conditions that can be accessed by Clicking on this link http://www.vodacom.net/legal/email.aspx "

1 0

[Serusers] SUBSCRIBE timeout again
by Andrey Kouprianov 13 Dec '05

13 Dec '05

Hi, I am having the same problem with SUBSCRIBEs again.... SER just keeps on sending me back timeouts for them... The problem persists only for any other presentity except myself. SUBSCRIBE timeouts happen most of the times, however, it's only sometimes, that they get through. Here's a part of ser.cfg responsible for SUBSCRIBE and NOTIFY: . . . . . . if(method=="SUBSCRIBE") { if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { log(1, "Authentication for SUBSCRIBE"); www_challenge("sip.interlab.ait.ac.th", "0"); break; }; if(!lookup("location") ) { log(1, "No registered user found for SUBSCRIBE request"); sl_send_reply("404", "Not Found"); break; }; }; if(method=="NOTIFY") { if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { log(1, "Authentication for SUBSCRIBE"); www_challenge("sip.interlab.ait.ac.th", "0"); break; }; if (!lookup("location")) { log(1, "No registered user found for NOTIFY request"); sl_send_reply("404", "Not Found"); break; }; }; if( !t_relay() ) { sl_reply_error(); }; . . . . . . . . Here are the traces: Message 1 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 1 SUBSCRIBE From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Content-Length: 0 Message 2 from 203.159.31.36:5060 to 203.159.32.39:5060 SIP/2.0 401 Unauthorized Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 1 SUBSCRIBE From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.5e75 Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 WWW-Authenticate: Digest realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0" Server: Sip EXpress router (0.9.3 (i386/freebsd)) Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=17754 req_src_ip=203.159.32.39 req_src_port=5060 in_uri=sip:12345@sip.interlab.ait.ac.th out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1" Content-Length: 0 Message 3 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 Message 4 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 . . . . . . And so it goes, until SER replies with Timeout. There are no error messages in the log. Can you tell me what is wrong???

1 0

[Serusers] include
by FaberK 13 Dec '05

13 Dec '05

Hi folks, I've a question. Is it possible to use "include" files into ser.cfg? I would like to keep the ser,cfg locked and use the include to make changes into routing. Any ideas? Thanks -- .:FaberK:.

2 1

RE: [Users] Can't create table './openser/pending.frm'
by Maria Yndefors 13 Dec '05

13 Dec '05

Yes, It seems the error was caused by too low number defined in MaxNoOfOrderedIndexes in the NDB cluster configuration. Thanks for the help. /Maria > -----Original Message----- > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org]On > Behalf Of Bogdan-Andrei Iancu > Sent: den 13 december 2005 12:25 > To: Maria Yndefors > Cc: users(a)openser.org > Subject: Re: [Users] Can't create table './openser/pending.frm' > > > Hi Maria , > > I would say it;s more a problem related to mysql setup - try > to google > to find the meaning of errno 4242 > > regards, > bogdan > > Maria Yndefors wrote: > > >Hallo, > >I have tried to install openser and run into this problem > when doing openser_mysql.sh create > > > >creating database openser ... > >ERROR 1005 (HY000) at line 215: Can't create table > './openser/pending.frm' (errno: 4242) > > > >I run Mysql server on another server and have set the > variables in the .opensermyslqrc file > > > >sbc2:/etc/openser# cat .opensermysqlrc > ># File .opensermysqlrc included by openser_mysql.sh > ># Set static variables to be used by the script > > > >export SQL_HOST=195.54.xxx.xxx > >export DBHOST=195.54.xxx.xxx > >export TABLE_TYPE="ENGINE=NDB" > >export SIP_DOMAIN=BREDBAND > > > >When I do show tables in mysql I can see: > > > >mysql> show tables; > >+-------------------+ > >| Tables_in_openser | > >+-------------------+ > >| acc | > >| active_sessions | > >| aliases | > >| dbaliases | > >| grp | > >| location | > >| missed_calls | > >| version | > >+-------------------+ > >8 rows in set (0.02 sec) > > > >some of the tables have been created. > >Why could not the pending table be created? > > > >Regards > >/Maria > > > > > > > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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