- sr-users - kamailio.org

[Serusers] Problems building RTPproxy on FC4
by sip 13 Dec '05

13 Dec '05

I'm having an issue building RTPproxy on FC4. I checked out the source yesterday, and the configure script went well, but the build is failing on some weird things. make all-am make[1]: Entering directory `/home/eneref/new/rtpproxy' gcc -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -c main.c main.c: In function 'handle_command': main.c:363: warning: pointer targets in passing argument 6 of 'recvfrom' differ in signedness main.c: In function 'main': main.c:997: error: 'useconds_t' undeclared (first use in this function) main.c:997: error: (Each undeclared identifier is reported only once main.c:997: error: for each function it appears in.) main.c:997: error: syntax error before 'delay' main.c:1230: error: 'delay' undeclared (first use in this function) make[1]: *** [main.o] Error 1 make[1]: Leaving directory `/home/eneref/new/rtpproxy' make: *** [all] Error 2 First off, useconds_t is in sys/types.h which is one of the includes in the main. But looking at the GCC line, it looks almost as though something's missing. I checked the Makefile, and the DEFAULT_INCLUDES is set to -I. -I$(srcdir) -I. and then there's that space... where I assume $(INCLUDES) would go... although that's not set anywhere in the Makefile. Anyone like to smack me with the bat of stupidity and tell me what I'm doing wrong? N.

1 0

[Serusers] Postgres instead of Mysql
by Felipe Louback 13 Dec '05

13 Dec '05

I am trying to use Postgres instead of Mysql, but I dont know why, then I initialize ser, I get the following error: 0(0) find_mod_export: <db_use_table> in module sql not found 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) ERROR: mod_init(): Can't bind database module I cant figure out the reason for this message "<db_use_table> in module sql not found". I have no idea why this is happening. When I was working with Mysql, I didn't have this problem, but now, I just cant put Postgres to work. Any ideas? Here is part of my ser.cfg: ============================================================ loadmodule "/usr/local/lib/ser/modules/postgres.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" # 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" modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc", "user_column", "username") modparam("usrloc", "db_url", "sql://ser:my_pass@localhost:5432/ser") =================================================================== Here is part of the log when I initialize SER: ============================================================= 0(0) DEBUG: init_mod: usrloc 0(0) usrloc - initializing 0(0) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(0) unixsock_register_cmd: New command (ul_stats) registered 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) find_mod_export: <db_use_table> in module sql not found 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) ERROR: mod_init(): Can't bind database module 0(0) init_mod(): Error while initializing module usrloc ERROR: error while initializing modules 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(27948) shm_mem_destroy 0(27948) destroying the shared memory lock ================================================================ Any clue is more than welcome... thanks. Felipe Louback -- Mestrando - Departamento de Engenharia Elétrica Grupo de Pesquisa em Engenharia da Computação e Telecomunicações Universidade Federal de Minas Gerais - Brasil "Todas as coisas cooperam para o bem daqueles que amam a Deus" Romanos 8:28

2 4

[Users] How to start ser on boot? how to monitor it?
by Cesc 13 Dec '05

13 Dec '05

Hi, An easy one ... or two. 1 - I need to start up ser automatically on the boot of the machine ... i am sure there are scripts and instructions on how to do it. Any pointer? 2 - I need a way to monitor the status of ser (and sems), so that in case of a crash they are automatically killed and restarted. I thought about writing some kind of cron thingy using serctl ... Again, pointers? Regards, Cesc

6 8

[Users] OpenSER 1.1
by Douglas Garstang 13 Dec '05

13 Dec '05

I see documentation available for OpenSER 1.1 at www.openser.org, but I can't seem to find anywhere where it's available for download. It is available yet? Thanks

3 2

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

13 Dec '05

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

2 1

[Serusers] Problems with STOP accounting
by Vesna Levkova Damjanovski On.net 13 Dec '05

13 Dec '05

We have a problem registering the stop accounting record in our database. We have freeradius server. When bye method occurs radius don't receive stop accounting record so we don't get the end time of call. The strange thing is that in some cases we received in some not.

1 0

[Serusers] Where does SER get the public IP of a NATed uac from for sending as STATUS message?
by Frank Fischer 13 Dec '05

13 Dec '05

Hi all related to 0.9.1pre. I have a problem with NATed UACs in my setup. Therefor i would like to know where SER takes the public IP of a NATed UAC from, a request is forwarded to (using the reply handler) if i call fix_nated_contact() to rewrite the NATed UACs contact header. Is it always taken form the received field in the location record? Just for explanation: It seems that my SER does not rewrite the contact header field in the reply route when i call fix_nated_contact(). The contact header value stays the same, just before and after call to fix_nated_contact(). So i think, it could be SER does not get the correct "NAT"-information for the related enpoint and the reads a value from a request field? Anyone ever encountered a similar situation? Is there a way to get closer information on what's happening when calling fix:nated_contact() during runtime? Thanks for you help. Kind regards Frank

4 4

[Serusers] post
by Vesna Levkova Damjanovski On.net 13 Dec '05

13 Dec '05

1 0

[Serusers] SER presence snapshot
by Vaclav Kubart 13 Dec '05

13 Dec '05

Hi all, there is new presence snapshot available at (simple-compilable sources with demo config): ftp://ftp.iptel.org/pub/ser/presence/ [it is now second version: ser-0.10.99-dev25-timers-pa-2_src.tar.gz] Actual version of "presence handbook" is there too: ftp://ftp.iptel.org/pub/ser/presence/presence_book.xhtml Vaclav

1 0

[Serusers] SER, presentity and mySQL
by Ángel Gómez 13 Dec '05

13 Dec '05

Hi, I'm working wich SER and making it to suport presence. Now I have SER instaled with the mySQL and pa modules loaded, the thing is than I make the pa module to use db to store the presence data, but when I send a Notify to the server its don't store any data in the presentity_contact table, and I want to know why it dosen't store anything in there. If some one knows why isn't stored any data, of if I make the notify bad, or if I doing this in the wrong way... please tell me, my objetive is to store the presentity status in teh data base. here is the notify than I use: NOTIFY sip:10.22.1.5:5060 SIP/2.0 To: <sip:angel@10.22.1.249> From: <sip:angel@10.22.1.249>;tag=edf6b60e8b292eeca4ee51d20e01ffa7-e36a CSeq: 1 NOTIFY Call-ID: af2756608d177822 Content-Length: 244 User-Agent: Sip EXpress router(0.9.3 (i386/linux)) Event: presence Content-Type: application/pidf+xml Subscription-State: active;expires=594 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "//IETF//DTD RFCxxxx PIDF 1.0//EN" "pidf.dtd"> <presence entity="sip:ninguna@10.22.1.249"> <tuple id="9r28r49">ser"> <status> <basic>closed</basic> </status> </tuple> </presence>

2 1

[Serusers] How can you check if From: SIP URI is Registered.
by Leo Papadopoulos 12 Dec '05

12 Dec '05

Dear SerUsers: I am running SER 0.9.4 and have just finished reading all the README's of all the modules, but cannot find what I am trying to do. I want to check all incoming messages to see if the SIP URI in the "From:" field is presently registered. I am using MySQL for registration and authentication. If the SIP URI is the "From:" field is NOT registerd I want to return a "403 Forbidden" message. I know that From: fields can be spoofed, but I want to do this anyway. Does anyone know of an existing module that can help with this? Or of anyway to do this? Leo P.

3 2

[Serusers] ERROR: cannot forward to af 2, proto 2 no corresponding listening socket
by Matt morris 12 Dec '05

12 Dec '05

Hello List, I got this error message when I try to do a forward_tcp: forward_req: ERROR: cannot forward to af 2, proto 2 no corresponding listening socket Do I need to configure SER to use tcp sockets at the top, or is it something else? Thank you all in advance. _________________________________________________________________ Take advantage of powerful junk e-mail filters built on patented Microsoft® SmartScreen Technology. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

1 0

[Serusers] SER don't start
by Guido 12 Dec '05

12 Dec '05

Hello, I'm using Fedora C4, ser-0.9.4 with radiusclient-ng-0.5.2. When I start SER I view this error message and SER don't start 0(14959) WARNING: fix_socket_list: could not rev. resolve 66.167.141.22 0(14959) WARNING: fix_socket_list: could not rev. resolve 66.167.141.22 Listening on udp: 66.165.171.22 [66.167.141.22]:5060 tcp: 66.165.171.22 [66.167.141.22]:5060 Aliases: WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) info: mediaproxy: loaded SIP asymmetric clients file containing 0 entries. 0(0) info: mediaproxy: loaded RTP asymmetric clients file containing 0 entries. 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 14961 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(14959) new_connection(): Access denied for user: 'serro@localhost' (Using password: YES) 0(14959) db_init(): Could not create a connection 0(14959) ERROR:domain_db_init: cannot initialize database connection 0(14959) ERROR: domain:child_init(): Unable to connect to the database 0(14959) init_mod_child(): Error while initializing module domain 0(14959) main_dontfork: init_child failed 1(14960) INFO: signal 15 received 2(14961) 0(14959) child process 14960 exited normally, status=0 0(14959) INFO: dont_fork turned on, living on INFO: signal 15 received 0(14959) child process 14961 exited normally, status=0 0(14959) INFO: dont_fork turned on, living on Can anybody help me ? Regards, Guido

2 1

[Serusers] problems to install ser with presence
by victoria beltran 12 Dec '05

12 Dec '05

Hi,I am trying to install the last unstable SER version with support for SIP-based presence. When I compile and install SER with presence modules (make install include_modules="pa rls mysql), it does this error: action.c: no such onsend.h route.c: no such onsend.h forward.c: no such onsend.h I have installed all libraries required (libxml2 y libcurl3) and the libxml2-dev and libcurl3-dev devel packages. The file "onsend.h" does not be in /sip_router directory. Could anybody help me? Thanks!! ______________________________________________ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es

3 4

[Serusers] ser_fifo and access denied issue (Maybee Context Problem) can you help?
by Douglas Mahy 12 Dec '05

12 Dec '05

I have had considerable success in getting ser working under selinux particularly as I am new to linux having just recently moved over from working in a windows environment. However I have had a lot of problems trying to resolve a permissions problem that is resulting in serweb kicking out the error. FIFO not running or bad path to it I have tried all the usual stuff such as including the lines I have also turned off selinux. fifo="/tmp/ser_fifo" fifo_mode=0777 or sock_mode=0666 I now believe that this may be something to do with context, is there anybody that has experienced this problem. Below is the error I am getting using dmseg, it increments each time I refresh the admin page. audit(1134382735.156:78): avc: denied { getattr } for pid=2682 comm="httpd" name="ser_fifo" dev=sda2 ino=1887848 scontext=root:system_r:httpd_t tcontext=root:object_r:tmp_t tclass=fifo_file Below are the permissions on the ser_fifo and its context. /dev/ brw-rw---- root disk system_u:object_r:fixed_disk_device_t sda1 brw-rw-rw- root disk system_u:object_r:fixed_disk_device_t sda2 brw-rw---- root disk system_u:object_r:fixed_disk_device_t sda3 /tmp/ -rw------- root root root:object_r:tmp_t php-install.GQ3113 prwxrwxrwx root root root:object_r:tmp_t ser_fifo prw-rw-rw- root root root:object_r:tmp_t ser_receiver_3011 Any help would be very much appreciated. Doug Mahy Telewest Broadband Head of Research & Development Network Engineering and Strategy M +44(0) 7985 807 534 Email: <mailto:douglas.mahy@telewest.co.uk> douglas.mahy(a)telewest.co.uk ------------------------------------------------------------------------------ Altogether Better www.telewest.co.uk The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. ==============================================================================

1 0

[Users] location table refresh
by Matteo Piazza 12 Dec '05

12 Dec '05

Hi all, I have a question. When I change the location tables is it necessary to restart openser? I see that if I don`t restart the openser the program don`t see the modifications on the location table. Is it possible to refresh without restart the program, which command can I use? Matteo -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 1

[Serusers] Problem configuring presence module
by Javi Lavandeira 12 Dec '05

12 Dec '05

Hello, I work for a Japanese company and I'm writing in behalf of a coworker who can read some English, but cannot write it. He tells me that he's trying to use authentication in SER's presence module, but gets the following error when trying to run it: Dec 2 16:55:22 gorilla3 ser: set_mod_param_regex: parameter <auth> not found in module <pa> Dec 2 16:55:22 gorilla3 ser: parse error (93,20-21): Can't set module parameter He thinks that the problem might be the following line in ser.cfg: modparam("pa", "auth", "xcap") The version he's using is ser-0.10.99 from CVS. Any idea on what could be the problem? Thanks in advance, -- Javi Lavandeira Technical manager International Systems Research http://www.isr.co.jp

2 1

Re: [Serusers] SER Presence
by Vaclav Kubart 12 Dec '05

12 Dec '05

These messages in log mean, that the SUBSCRIBE request didn't contain any known document types in Accept headers. There was a bug in PA module - it processed only the first Accept header, it was corrected on Friday, thus with actual CVS version it may work; but only in the case, that your client sends more Accept headers and one of them is known to PA module. If it will NOT work with actual version, please send me the message flow. thanks, Vaclav

1 0

[Serusers] problems to install SER with presence
by victoria beltran 12 Dec '05

12 Dec '05

Hi, I am trying to install the last unstable SER version with support for SIP-based presence. When I compile and install SER with presence modules (make install include_modules="pa rls mysql), it does several errors: receive.o: In the function 'receive_msg': /root/sip_router/receive.c:231:undefined reference to 'reset_user_avps' receive.o:/root/sip_router/receive.c:232:undefined reference to 'reset_user_avps' receive.o:/root/sip_router/receive.c:251:undefined reference to 'reset_user_avps' receive.o:/root/sip_router/receive.c:252:undefined reference to 'reset_domain_avps' I have downloaded current ser sources by "cvs -d :pserver:anonymous@cvs.berlios.de:/cvsroot/ser checkout sip_router" and I have seen the usr_avp.h and user_avp.c files. In this files, the reset_user_avps and reset_domain_avps functions hasnt been defined. Could anybody help me? Thanks!! --------------------------------- Correo Yahoo! Comprueba qué es nuevo, aquí http://correo.yahoo.es

3 2

[Users] Routing
by Douglas Garstang 12 Dec '05

12 Dec '05

We have a configuration where OpenSER sit's between Polycom phones and Asterisk. Off-net/PSTN calls are also sent from Asterisk to OpenSER to be routed to a Mediacodes Gateway. I'm having a hell of a time with the routing in general! What is the best way to route calls? Should I be looking at the URI? The From:? The To:? The method? Also, the convention seems to be that the URI contains where the message is going TO rather than where it came FROM. However, routing is much easier if the URI has where it came from. For example, messages from the phones have a URI of: uac.voip.com and messages from Asterisk have the IP adress in the URI, as Asterisk doesn't seem to let you set the URI. Calls from the Mediacodes have a URI of: pstn.voip.com Is adding sub-domains like 'uac' and 'pstn' a good way to route? I've looked at all the examples I could find but unfortunately they're ridiculously simple. Every time I look at openser.cfg I get VERY STRESSED OUT! Thanks.

2 1

[Users] 302 and contact address new invites an request header..
by Helge Waastad 12 Dec '05

12 Dec '05

Hi, I was just wondering if there has been any changes to the pseudo variables? I receive a 302 with a new contact address, however if I print out in the failure router $ct I get the $from..... br hw PS, another thing I've realized now, is that I have a problem solving the new INVITE after a redirect. Since the new INVITE after redirect has the same call-id, does that mean that all flags etc is kept and still are valid? For instance; INVITE: xxx@<cisco-gw> -> yyy@<domain> REDIRECT: <- 302 Contact: zzz@<domain> NEW INVITE xxx@<cisco-gw> -> zzz@<domain> I have a : if (uri=~"^sip:yyy@.*") in the beginning of my script and it return true on the new INVITE.. Is this right? br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 3

Fwd: [Serusers] Adding users in the loaded db
by samuel 12 Dec '05

12 Dec '05

Forgot to send it to the list.....I'm still on holiday mood..... Sam. ---------- Forwarded message ---------- From: samuel <samu60(a)gmail.com> Date: 12/12/2005 09:59 Subject: Re: [Serusers] Adding users in the loaded db To: Xavier TRENTIN <xtrentin(a)softiphone.net> Hi all!!! If you use serctl, it will do both load the user in the database and in the cache automatically. what serctl does: Update the database: standard [mysql] query. Update SER's cache: it makes use of the exported FIFO command ul_add: serctl fifo ul_add <table> <aor> <contact> <expires> <q> <replicat> <flags> example: serctl fifo ul_add location sam(a)example.com sam(a)1.2.3.4 3600 1 0 0 about the dbtext...let my guess...I've been out for a while and things may have changed but I think serctl in the SER CVS does not handle dbtext...there was a post from Cesc with some patche adding dbtext support to serctl and I think that in the openser repository there's some dbtext serctl.... Hope it helps, Sam. 2005/12/11, Xavier TRENTIN <xtrentin(a)softiphone.net>: > I agree with you except that if SER is running and I add a user in my > dbtext subscriber table, > he can't register if I don't restart SER. > > sip a écrit : > > >*********************** > >Pas de virus detecté dans l'attachement no filename > > > >Votre Mail a été vérifié par InterScan. > >***********-*********** > > > > > >SER doesn't really load users on startup into memory except from the location > >table (i.e. user/contact information). When you create a user in the > >subscriber table, that user is added. Then, when his client sends a register, > >he gets added to the location table in memory/in the db. There's nothing you > >really need to to to manually add someone to the location table... and really, > >there's nothing you SHOULD be doing. Users add themselves on login with all > >the correct information about their contact info. > > > >N. > > > > > >On Fri, 09 Dec 2005 18:58:56 +0100, Xavier TRENTIN wrote > > > > > >>Hi, > >> > >>I'd like to know if there is a way to add users in the database > >>loaded in memory, using a fifo command for example? > >> > >>If, yes, could you give an exemple. > >> > >>Cause I don't want to restart ser anytime I add a user ;p > >> > >>B.R. > >>Xavier. > >> > >>_______________________________________________ > >>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 >

1 0

[Serusers] SER Presence
by Tina Suneli 12 Dec '05

12 Dec '05

Hi, I have just joined that list and want to know why the following error is occuring when i used to subscribe with SER with an OPAL 2.0 client. 0(2876) handle_subscription() entered 0(2876) check_message(): Accepts application/simple-message-summary not valid for event package et=message-summary 0(2876) handle_subscription(): Error while checking message 0(2876) handle_subscription about to send_reply and return -2 Tina

1 0

[Serusers] SER and REGISTRATIONs UAC
by Federico Giannici 11 Dec '05

11 Dec '05

We use SER as a proxy to a PSTN gateway that is "transparent" to the final users: the users register on the proxy and it authenticate to the gateway (by means of the UAC module). Currently we use Asterisk for the registration on the PSTN gateway, but we'd like to get rid of it. What is the best way to use SER to register on the gateway? We have to write a little program that sends REGISTER messages to the gateway, or somebody already developed something (maybe a module)? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 2

[Serusers] Adding users in the loaded db
by Xavier TRENTIN 11 Dec '05

11 Dec '05

Hi, I'd like to know if there is a way to add users in the database loaded in memory, using a fifo command for example? If, yes, could you give an exemple. Cause I don't want to restart ser anytime I add a user ;p B.R. Xavier.

2 3

[Serusers] http tunneling
by Imran Akbar 10 Dec '05

10 Dec '05

Hi, I've tried to use a softphone from behind a corporate firewall which only allows http requests through its proxy - and it doesn't work, even with STUN. I've heard that tunneling through HTTP would allow VoIP to work in this situation - does anyone know how to set this up, with SER or other hardware/software? Thanks, Imran

3 2

[Serusers] SER + SEMS & Music on Hold
by Waldo Rubinstein 09 Dec '05

09 Dec '05

Does anyone have any configuration sample on setting up SER + SEMS in order to "put a call on hold and listen to MoH" while SER reaches the destination peer(s)? Or at least point to some documentation referencing this? Thanks, Waldo

1 0

[Serusers] problems with ser + radius
by Guido 09 Dec '05

09 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 2

[Serusers] announcement: student project finished: partiSIPation
by Juha Heinanen 09 Dec '05

09 Dec '05

matthias, i looked at partiSIPpation documentation and didn't quite understand why microphone and speaker volumes are part of the api. i would think that those are internal matters of the gui component. when gui places a call, i would imagine it gives to core an ip address and port that it wants to use for the media plus a list of codecs it supports. i don't understand what volume issue have to do with communication between gui and core. -- juha

2 3

[Users] multi-leg accounting
by Helge Waastad 09 Dec '05

09 Dec '05

Hi, to get multi-leg accounting to work, is it necessary to avp_write "something" to the src_leg_avp_id/dst_leg_avp_id avp's? Right now I only get "n/a" in those columns. I've tried to use avp_write without any success. br, hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

3 5

[Users] RE: [Devel] unixODBC Database Connection Problem
by Lenir 09 Dec '05

09 Dec '05

Ok, I tried using the FreeTDS driver for unixodbc and I got a more detailed error message that might shine a light: Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect IM007:1:0:[unixODBC][FreeTDS][SQL Server]Could not find UID parameter HYC00:2:0:[unixODBC][FreeTDS][SQL Server]Driver not capable 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration These are my entries in /etc/odbc.ini [SYBASE] Driver = /usr/local/lib/libtdsodbc.so Description = Sybase Server Trace = Yes Trafcefile = /tmp/odbc.log #Servername = JDBC Server = xx.xx.xx.xx Port = 5000 DSN = database Database = database #UserID = username # I tried using both UID and UserID, had same results UID = username Password = passwd TDS_Version = 5 These are my entries in /etc/freetds.conf: [SYBASE] host = xx.xx.xx.xx port = 5000 tds version = 5 For whatever reason the module is not picking up the UID and PWD from the db_url parameter i.e. modparam("usrloc|auth_db", "db_url", "unixodbc://username:passwd@xx.xx.xx.xx/SYBASE") The way I fixed the problem was by changing the file my_con.c, line 69: From: sprintf( stringDNS, "%s%s%s", "DSN=", id->database, ";"); to: sprintf( stringDNS, "%s%s%s", "DSN=", id->database, ";UID=username;PWD=passwd;"); And now works fine. Also, to add debug to figure out which SQL queries fail, I changed the file dbase.c line 75: From: LOG(L_ERR, "Return value: %d\n", ret); To: LOG(L_ERR, "Return value: %d, Performing query: %s\n", ret, _s); Now everytime a query fails it'll say the result msg, as well as the actual query that failed. Lenir _____ From: devel-bounces(a)openser.org [mailto:devel-bounces@openser.org] On Behalf Of lenirsantiago(a)yahoo.com Sent: Thursday, December 08, 2005 12:59 PM To: devel(a)openser.org; users(a)openser.org Subject: [Devel] unixODBC Database Connection Problem Guys, Im trying to use unixodbc.so to connect to a Sybase database using Sybase native ODBC drivers. This is the entry that I have on: /etc/odbc.ini: ============= [SYBASE] Description = Sybase ODBC Data Source UserID = username Password = passwd Driver = ASE Server = xx.xx.xx.xx Port = 5000 Database = database UseCursor = 1 /etc/odbcinst.ini ============ [ASE] Description = Sybase ODBC Driver Driver = /opt/sybase/ODBC-12_5/lib/libsybdrvodb.so FileUsage = -1 UsageCount = 1 The database is on a remote host. I can connect from the openser server using unixodbc's isql tool, as such: [root@ser openser]# isql SYBASE username passwd +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> This proves that unixodbc is installed correctly and that the Sybase odbc driver is installed and functioning properly. However when I try to run openser (debug=4) I get the following: 0(0) fixing /usr/local/lib/openser/modules/auth_db.so www_authorize 0(0) db_init: Connection 'unixodbc://username:passwd@xx.xx.xx.xx/SYBASE' not found in pool Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(22871) shm_mem_destroy 0(22871) destroying the shared memory lock [root@ser openser]# This is a snippet of my openser.cfg ------------------------------START-------------------------------- debug=4 fork=no log_stderror=yes listen=yy.yy.yy.yy port=5060 children=4 server_signature=no dns=no rev_dns=no check_via=no syn_branch=yes memlog=1000 sip_warning=no fifo_mode=0666 fifo="/tmp/openser_fifo" fifo_db_url="unixodbc://username:passwd@xx.xx.xx.xx/SYBASE" reply_to_via=no loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/postgres.so" loadmodule "/usr/local/lib/openser/modules/unixodbc.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/acc.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_radius.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/domain.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/options.so" #loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" modparam("usrloc", "db_mode", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("usrloc|auth_db", "db_url", "unixodbc://username:passwd@xx.xx.xx.xx/SYBASE") ---------------------------------END---------------------------------------- - Any suggestions? Lenir

1 0

[Users] Relaying and failure route - misrouted ACK
by Stefan Prelle 09 Dec '05

09 Dec '05

Hi folks, I need an advice regarding failure routes. I want the (Open)SER to route all incoming calls to a PSTN gateway. If that route fails (e.g. because the gateway is down) it shall route the calls to another gateway. I managed to archieve that in the failure case the second gateway receives the call. But the ACK request that the caller sends when receiving the 200 OK is delivered to the primary target again, not to the fallback gateway. My configuration (excerpt): route { ... record_route(); if (loose_route()) { t_relay(); exit; }; t_on_failure("1"); t_relay_udp("192.168.0.37","5060"); } failure_route[1] { append_branch(); t_relay_to_udp("192.168.0.39", "5060"); } Thanks in advance. Regards, Stefan

2 6

[Serusers] Radius - LDAP
by Paco Orozco 09 Dec '05

09 Dec '05

Hiya, I'm trying to use RADIUS server to authorize (authenticate) SIP users. I'm using auth_radius module. My radius server is the same machine, FreeRadius wich is configured to use LDAP as DB. Whe I try to authenticate a SIP user I've got a problem, SIP SER sends a radius request with digest method, but my Radius needs a User-Password suplied in order to bind in LDAP. How can intruct SIP SER in order to pass auth password as User-Password althout Digest??? Thanks in advance. -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida.

2 2

[Users] New Feature: Supported methods and lookup
by Bogdan-Andrei Iancu 09 Dec '05

09 Dec '05

Hi, based on Juha's patch - saving in usrloc the supported methods advertised during REGISTER - I extended lookup() function to perform method based filtering. What this means: lookup() may return only contacts which supports the method of the processed request. Ex: there are two clients registering as same user: 1) a SIP phone which support INVITE & family, but no presence and messaging. 2) a soft messaging client : supports only MESSAGE When a call hits the server, the lookup() will send it only to contact (1), to the phone, since the contact (2) does not support INVITE. When a MESSAGE hits the server, the lookup() will send it only to contact (2), to the soft client, since the contact (1) does not support MESSAGE. the advantage is that you avoid un-necessary parallel forking and extra signalling. Also there are SIP implementation which does not even reply to methods they are not supporting, forcing the proxy to generate timeout..:( The feature may be turned on or off via the module parameter "method_filtering": http://www.openser.org/docs/modules/1.1.x/registrar.html#AEN247 For more, see also the script snapshot from : http://www.openser.org/docs/modules/1.1.x/registrar.html#AEN288 regards, bogdan

2 4

RE: [Serusers] What does SER/FIFO error usually mean?
by Yong, Chee Yeew 09 Dec '05

09 Dec '05

That worked. Thanks. I am not sure why this (fifo_mode=0777) wasn't inserted in either the ser.cfg or ser.cfg.sample files. Chee -----Original Message----- From: Steve Blair [mailto:blairs@isc.upenn.edu] Sent: 08 December 2005 18:39 To: Yong, Chee Yeew Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] What does SER/FIFO error usually mean? Permissions problem I'm guessing. Do you have the following in your config file? fifo="/tmp/ser_fifo" # named pipe needed by serctl fifo_mode = 0777 Yong, Chee Yeew wrote: > Hello all, > > I am trying to add a user to my MySql database with the command: > > serctl add joeuser passwd joeuser(a)here.com <mailto:joeuser@here.com> > > but I am getting an error: > error: SER/FIFO not accessible: 2 > > Any pointers to where to start debuging this error? > > Thanks. > > chee > >------------------------------------------------------------------------ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

2 1

Re: [Serusers] Set up aliases
by Anthony Chapellier 09 Dec '05

09 Dec '05

It doesn't work. It seems 'mhomed = 1' binds ser to localhost. So when I try to forward the request to my SIP server after I did the local forward to save the user alias, it says 'can't access to the net'. Anthony Michal Matyska a écrit : >Try to add following line into your config file: >mhomed=1 > >Michal > >On Fri, Dec 09, 2005 at 01:46:26PM +0100, Anthony Chapellier wrote: > > >>I'm using SER 0.9.4. >> >>Michal Matyska a écrit : >> >> >> >>>Which version did you use? >>>I'm getting the same error... looking into the code, but nothing found yet. >>> >>>Michal >>> >>>On Fri, Dec 09, 2005 at 12:19:20PM +0100, Anthony Chapellier wrote: >>> >>> >>> >>> >>>>I've got a problem with forwarding in loopback. I tried to forward the >>>>request with the following code (tested both) : >>>> >>>>t_relay_to_udp("127.0.0.1", "5060"); >>>> >>>>or >>>> >>>>forward(127.0.0.1, 5060); >>>> >>>>In both of them, I got an error 477 : Can't process to the next hop - >>>>127.0.0.1:5060 Noisy Feefback... Do you know where it could come from ? >>>> >>>>Thanks, >>>> >>>>Anthony >>>> >>>>Jan Janak a écrit : >>>> >>>> >>>> >>>> >>>> >>>>>Forward the request to the same proxy again over loopback. In the 2nd >>>>>run you can detect that the request is coming from 127.0.0.1 and save it >>>>>in aliases table instead of location. >>>>> >>>>>jan. >>>>> >>>>>On 08-12-2005 11:15, Anthony Chapellier wrote: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>>Hi, >>>>>> >>>>>>When a register request is incoming, I save user location and rewrite >>>>>>contact header after that. I'd like to save this new rewriten user >>>>>>location as an alias, is there a way to set up an alias in the fly ? >>>>>> >>>>>>I saw you need to use serctl command to add aliases but can you do it >>>>>>in SER in real time ? >>>>>> >>>>>>Thanks, >>>>>> >>>>>>_______________________________________________ >>>>>>Serusers mailing list >>>>>>serusers(a)lists.iptel.org >>>>>>http://lists.iptel.org/mailman/listinfo/serusers >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> >>>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> >>> >>> >>> >>> > > > > > >

1 0

[Serusers] Jain SIMPLE
by Andrey Kouprianov 09 Dec '05

09 Dec '05

I know this is not the proper place to ask this kind of question, but does anybody have any idea, when JAIN SIMPLE package is coming out?

2 1

[Serusers] Set up aliases
by Anthony Chapellier 09 Dec '05

09 Dec '05

Hi, When a register request is incoming, I save user location and rewrite contact header after that. I'd like to save this new rewriten user location as an alias, is there a way to set up an alias in the fly ? I saw you need to use serctl command to add aliases but can you do it in SER in real time ? Thanks,

3 4

[Serusers] create user problem with serweb
by Ashutosh kumar 09 Dec '05

09 Dec '05

Hey, We have worked up our own installation of serweb without Confirmation mail by tweaking the sql. What *ROUGHLY* you have to do is to 1. remove the "sql insert" into pending table from reg/index.php. 2. remove the "sql table copy" (select * into from...) from pending to subscribers table in reg/confirmation.php. 3. Add the equivalent of above two codes as a single insert into subscribers in the file reg/index.php. (I may be missing the file names correctly though, since I did that a long ago.) This will cause the registration records to directly go into the subscribers table without going to the pending table. You might want to change/edit the related code and messages accordingly. But remember, it is always good to follow the prescribed path. Regards, Ashutosh Kumar -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Karel Kozlik Sent: Friday, December 09, 2005 1:47 PM To: ciprian niculescu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] create user problem with serweb Hi, unfotunately current version of serweb doesn't allow to register users without sending confirmation mail. It is feature of planed next release. I hope the beta version will be avaiable shortly after New Year. You may use serctl for now. regards, Karel ciprian niculescu napsal(a): > hi, > > i have installed ser and serweb, i use it for teaching sip signaling, > and i don't want to use confirmation mail for creating user accounts, > i've looked around the docs/code but nothing pop up. > > i'm using CentOS, mysql, last ser and last serweb > > ciprian > > _______________________________________________ > 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

1 0

Re: [Users] transfer not possible!!
by Jayesh Nambiar 09 Dec '05

09 Dec '05

hi klaus, Thanx for your response. I have hard phones connected to Audio codes 4-port gateway. The gateway is configured to send the calls to openser for processing. suppose there are three users connected to it. user1 calls user2. The call goes through. Now user 2 presses flash and tries to call user3, but both the phone gets disconnected. I did an ngrep and what i noticed was this: When user2 tries to transfer the call to user 3, the "INVITE" goes from user2 to user1 again and the call gets disconnected due to request timeout. I am not able to see any REFER method in the ngrep. Any advice will be appreciated. Thanx again. Jayesh Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: Yes, REFER method is for call transfer. Which clients do you use? Does the call orginate also from a SIP phone or from a gateway? What exactly is the problem? Do you see the REFER request? Including an "ngrep port 5060" would be useful. klaus Jayesh Nambiar wrote: > hi everyone, > > I was just trying to transfer the call to another user. All users are > registered at the same openser server. Now when I receive a call, is it > not possible to transfer the call to another IP user. > I thought REFER mehtod does this, but i think I am wrong. Is this > possible, can somebody pls clear my understanding regarding this concept. > > thanks in advance!! > jayesh > > ------------------------------------------------------------------------ > Enjoy this Diwali with Y! India Click here > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users --------------------------------- Yahoo! India Matrimony: Find your partner now.

3 5

[Serusers] mysql.so
by rajesh.rajguru＠wipro.com 09 Dec '05

09 Dec '05

Hi, I have downloaded ser-0.9.4_src.tar . After compiling we found that mysql.so is not created. When I try to compile this module separately it gives error that mysql.h file is not present and when I check in modules/mysql, it is not present. How to build mysql.so module then ? Thanks in advance for the help Regards, Rajesh Confidentiality Notice 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 confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

3 3

[Users] Re: unixODBC Database Connection Problem
by Marco Lorrai 09 Dec '05

09 Dec '05

If you cannot connect to database using the command isql (provided by unixodbc) the module cannot work. You should first verify the command isql work. A problem could be a wrong connection socket, if is this restart Sysbase with the correct socket file. -- Marco Lorrai - marco.lorrai(a)abbeynet.it R&D Software Engineering Abbeynet S.p.A. - http://www.abbeynet.it VoIP (Office): +39-070-2339262 Fax: +39-070-229016

1 0

[Serusers] create user problem with serweb
by ciprian niculescu 09 Dec '05

09 Dec '05

hi, i have installed ser and serweb, i use it for teaching sip signaling, and i don't want to use confirmation mail for creating user accounts, i've looked around the docs/code but nothing pop up. i'm using CentOS, mysql, last ser and last serweb ciprian

2 1

[Serusers] Rtpproxy & Nathelper config
by Joel Solanki 09 Dec '05

09 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 3

[Serusers] Development of Berkeley DB module for SER
by Jiri Kuthan 09 Dec '05

09 Dec '05

I am seeking SER programmers for development of Berkeley DB driver for SER under iptel's sponsorship. I'm inviting all folks interested in this activity to contact me privately. I will prefer applicants who send me samples of their SER code in attachement. Sanity and performance of the resulting code is important to me. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] Help!!
by George Lambson 09 Dec '05

09 Dec '05

Does anyone know what I would put in my ser.cfg to authenticate by IP address? and have mediaproxy still handle that interaction? Thanks, George

1 0

[Serusers] announcement: student project finished: partiSIPation
by Matthias Liebig 09 Dec '05

09 Dec '05

Hello, recently we, a group of students, finished our project here at iptel. It is a SIP user agent framework. Below you'll find the introduction of our project report which is published at BerliOS [1]. "PartiSIPation is the result of a student project at the Institute for Open Communication Systems (OKS) at the Technical University Berlin. When we started with PartiSIPation in April 2005 there already were some SIP clients available, but none of them was actually usable. So our main goal was to create a SIP soft client that does not have to hide itself from hard clients - SIP phones. Therefore it has to be able to handle several parallel calls and accounts. But we also thought about future applications like online games. So PartiSIPation should be easy integrateable in other applications on different operating systems. To reach all these goals we designed PartiSIPation as follows: PartiSIPation is a SIP user agent which consists of two parts: a Core written in portable C and an easily replaceable GUI. So it can be used as a softphone as well as it can be integrated in any kind of application. It uses external libraries for signaling and speech transfer which are called over interfaces and adapters. An adapter encapsulates the interface of a library and provides a simpler interface expected by the user agent. This makes an easy library replacement possible. Only the adapter has to be edited, no Core changes are necessary. Because PartiSIPation comes with its own SIP state machine a GUI or application can easily place calls by using its simple XML-RPC API. XML-RPC is language independent which means that the GUI can base on Java, GTK, Qt or whatever depending on the preferences of the GUI developer or its purpose. Every call has its own state machine, so parallel calls are no problem. And together with the account management of PartSIPation which allows the registration with several accounts it is possible to use this user agent for complex situations like they are occurring in gaming. Recently, PartiSIPation became Open Source Software and was published at BerliOS: http://developer.berlios.de/projects/partisipation/ ." Best regards, Matthias Liebig [1] http://developer.berlios.de/docman/?group_id=5063

1 0

[Users] RE: [Devel] unixODBC Database Connection Problem
by Lenir 08 Dec '05

08 Dec '05

Ok, I tried using the FreeTDS driver for unixodbc and I got a more detailed error message that might shine a light: Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect IM007:1:0:[unixODBC][FreeTDS][SQL Server]Could not find UID parameter HYC00:2:0:[unixODBC][FreeTDS][SQL Server]Driver not capable 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration These are my entries in /etc/odbc.ini [SYBASE] Driver = /usr/local/lib/libtdsodbc.so Description = Sybase Server Trace = Yes Trafcefile = /tmp/odbc.log #Servername = JDBC Server = xx.xx.xx.xx Port = 2000 DSN = database Database = database #UserID = username # I tried using both UID and UserID, had same results UID = username Password = passwd TDS_Version = 4.2 These are my entries in /etc/freetds.conf: [SYBASE] host = xx.xx.xx.xx port = 2000 tds version = 4.2 _____ From: devel-bounces(a)openser.org [mailto:devel-bounces@openser.org] On Behalf Of lenirsantiago(a)yahoo.com Sent: Thursday, December 08, 2005 12:59 PM To: devel(a)openser.org; users(a)openser.org Subject: [Devel] unixODBC Database Connection Problem Guys, Im trying to use unixodbc.so to connect to a Sybase database using Sybase native ODBC drivers. This is the entry that I have on: /etc/odbc.ini: ============= [SYBASE] Description = Sybase ODBC Data Source UserID = username Password = passwd Driver = ASE Server = xx.xx.xx.xx Port = 5000 Database = database UseCursor = 1 /etc/odbcinst.ini ============ [ASE] Description = Sybase ODBC Driver Driver = /opt/sybase/ODBC-12_5/lib/libsybdrvodb.so FileUsage = -1 UsageCount = 1 The database is on a remote host. I can connect from the openser server using unixodbc's isql tool, as such: [root@ser openser]# isql SYBASE username passwd +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> This proves that unixodbc is installed correctly and that the Sybase odbc driver is installed and functioning properly. However when I try to run openser (debug=4) I get the following: 0(0) fixing /usr/local/lib/openser/modules/auth_db.so www_authorize 0(0) db_init: Connection 'unixodbc://username:passwd@xx.xx.xx.xx/SYBASE' not found in pool Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(22871) shm_mem_destroy 0(22871) destroying the shared memory lock [root@ser openser]# This is a snippet of my openser.cfg ------------------------------START-------------------------------- debug=4 fork=no log_stderror=yes listen=yy.yy.yy.yy port=5060 children=4 server_signature=no dns=no rev_dns=no check_via=no syn_branch=yes memlog=1000 sip_warning=no fifo_mode=0666 fifo="/tmp/openser_fifo" fifo_db_url="unixodbc://username:passwd@xx.xx.xx.xx/SYBASE" reply_to_via=no loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/postgres.so" loadmodule "/usr/local/lib/openser/modules/unixodbc.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/acc.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_radius.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/domain.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/options.so" #loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" modparam("usrloc", "db_mode", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("usrloc|auth_db", "db_url", "unixodbc://username:passwd@xx.xx.xx.xx/SYBASE") ---------------------------------END---------------------------------------- - Any suggestions? Lenir

1 0

[Serusers] What does SER/FIFO error usually mean?
by Yong, Chee Yeew 08 Dec '05

08 Dec '05

Hello all, I am trying to add a user to my MySql database with the command: serctl add joeuser passwd joeuser(a)here.com <mailto:joeuser@here.com> but I am getting an error: error: SER/FIFO not accessible: 2 Any pointers to where to start debuging this error? Thanks. chee

2 1

[Users] unixODBC Database Connection Problem
by lenirsantiago＠yahoo.com 08 Dec '05

08 Dec '05

Guys, Im trying to use unixodbc.so to connect to a Sybase database using Sybase native ODBC drivers. This is the entry that I have on: /etc/odbc.ini: ============= [SYBASE] Description = Sybase ODBC Data Source UserID = username Password = passwd Driver = ASE Server = xx.xx.xx.xx Port = 5000 Database = database UseCursor = 1 /etc/odbcinst.ini ============ [ASE] Description = Sybase ODBC Driver Driver = /opt/sybase/ODBC-12_5/lib/libsybdrvodb.so FileUsage = -1 UsageCount = 1 The database is on a remote host. I can connect from the openser server using unixodbc's isql tool, as such: [root@ser openser]# isql SYBASE username passwd +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> This proves that unixodbc is installed correctly and that the Sybase odbc driver is installed and functioning properly. However when I try to run openser (debug=4) I get the following: 0(0) fixing /usr/local/lib/openser/modules/auth_db.so www_authorize 0(0) db_init: Connection 'unixodbc://username:passwd@xx.xx.xx.xx/SYBASE' not found in pool Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(22871) shm_mem_destroy 0(22871) destroying the shared memory lock [root@ser openser]# This is a snippet of my openser.cfg ------------------------------START-------------------------------- debug=4 fork=no log_stderror=yes listen=yy.yy.yy.yy port=5060 children=4 server_signature=no dns=no rev_dns=no check_via=no syn_branch=yes memlog=1000 sip_warning=no fifo_mode=0666 fifo="/tmp/openser_fifo" fifo_db_url="unixodbc://username:passwd@xx.xx.xx.xx/SYBASE" reply_to_via=no loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/postgres.so" loadmodule "/usr/local/lib/openser/modules/unixodbc.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/acc.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_radius.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/domain.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/options.so" #loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" modparam("usrloc", "db_mode", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("usrloc|auth_db", "db_url", "unixodbc://username:passwd@xx.xx.xx.xx/SYBASE") ---------------------------------END---------------------------------------- - Any suggestions? Lenir

1 0

[Users] RE: [Devel] unixodbc connection problem
by Lenir 08 Dec '05

08 Dec '05

Ok, I tried using the FreeTDS driver for unixodbc and I got a more detailed error message that might shine a light: Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect IM007:1:0:[unixODBC][FreeTDS][SQL Server]Could not find UID parameter HYC00:2:0:[unixODBC][FreeTDS][SQL Server]Driver not capable 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration These are my entries in /etc/odbc.ini [SYBASE] Driver = /usr/local/lib/libtdsodbc.so Description = Sybase Server Trace = Yes Trafcefile = /tmp/odbc.log #Servername = JDBC Server = xx.xx.xx.xx Port = 2000 DSN = database Database = database #UserID = username # I tried using both UID and UserID, had same results UID = username Password = passwd TDS_Version = 4.2 These are my entries in /etc/freetds.conf: [SYBASE] host = xx.xx.xx.xx port = 2000 tds version = 4.2 _____ From: devel-bounces(a)openser.org [mailto:devel-bounces@openser.org] On Behalf Of Lenir Santiago Sent: Thursday, December 08, 2005 11:18 AM To: devel(a)openser.org; users(a)openser.org Subject: [Devel] unixodbc connection problem Guys, Im trying to use unixodbc.so to connect to a Sybase database using Sybase native ODBC drivers. This is the entry that I have on: /etc/odbc.ini: ============= [SYBASE] Description = Sybase ODBC Data Source UserID = username Password = passwd Driver = ASE Server = xx.xx.xx.xx Port = 5000 Database = database UseCursor = 1 /etc/odbcinst.ini ============ [ASE] Description = Sybase ODBC Driver Driver = /opt/sybase/ODBC-12_5/lib/libsybdrvodb.so FileUsage = -1 UsageCount = 1 The database is on a remote host. I can connect from the openser server using unixodbc's isql tool, as such: [root@ser openser]# isql SYBASE username passwd +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> This proves that unixodbc is installed correctly and that the Sybase odbc driver is installed and functioning properly. However when I try to run openser (debug=4) I get the following: 0(0) fixing /usr/local/lib/openser/modules/auth_db.so www_authorize 0(0) db_init: Connection 'unixodbc://username:passwd@xx.xx.xx.xx/SYBASE' not found in pool Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(22871) shm_mem_destroy 0(22871) destroying the shared memory lock [root@ser openser]# This is a snippet of my openser.cfg ------------------------------START-------------------------------- debug=4 fork=no log_stderror=yes listen=yy.yy.yy.yy port=5060 children=4 server_signature=no dns=no rev_dns=no check_via=no syn_branch=yes memlog=1000 sip_warning=no fifo_mode=0666 fifo="/tmp/openser_fifo" fifo_db_url="unixodbc://username:passwd@xx.xx.xx.xx/SYBASE" reply_to_via=no loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/postgres.so" loadmodule "/usr/local/lib/openser/modules/unixodbc.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/acc.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_radius.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/domain.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/options.so" #loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" modparam("usrloc", "db_mode", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("usrloc|auth_db", "db_url", "unixodbc://username:passwd@xx.xx.xx.xx/SYBASE") ---------------------------------END---------------------------------------- - Any suggestions? Lenir

1 0

[Users] unixodbc connection problem
by Lenir Santiago 08 Dec '05

08 Dec '05

Guys, Im trying to use unixodbc.so to connect to a Sybase database using Sybase native ODBC drivers. This is the entry that I have on: /etc/odbc.ini: ============= [SYBASE] Description = Sybase ODBC Data Source UserID = username Password = passwd Driver = ASE Server = xx.xx.xx.xx Port = 5000 Database = database UseCursor = 1 /etc/odbcinst.ini ============ [ASE] Description = Sybase ODBC Driver Driver = /opt/sybase/ODBC-12_5/lib/libsybdrvodb.so FileUsage = -1 UsageCount = 1 The database is on a remote host. I can connect from the openser server using unixodbc's isql tool, as such: [root@ser openser]# isql SYBASE username passwd +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> This proves that unixodbc is installed correctly and that the Sybase odbc driver is installed and functioning properly. However when I try to run openser (debug=4) I get the following: 0(0) fixing /usr/local/lib/openser/modules/auth_db.so www_authorize 0(0) db_init: Connection 'unixodbc://manager:purav1da@216.82.124.34/SYBASE' not found in pool Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(22871) shm_mem_destroy 0(22871) destroying the shared memory lock [root@ser openser]# This is a snippet of my openser.cfg ------------------------------START-------------------------------- debug=4 fork=no log_stderror=yes listen=yy.yy.yy.yy port=5060 children=4 server_signature=no dns=no rev_dns=no check_via=no syn_branch=yes memlog=1000 sip_warning=no fifo_mode=0666 fifo="/tmp/openser_fifo" fifo_db_url="unixodbc://username:passwd@xx.xx.xx.xx/SYBASE" reply_to_via=no loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/postgres.so" loadmodule "/usr/local/lib/openser/modules/unixodbc.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/acc.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_radius.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/domain.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/options.so" #loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" modparam("usrloc", "db_mode", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("usrloc|auth_db", "db_url", "unixodbc://username:passwd@xx.xx.xx.xx/SYBASE") ---------------------------------END---------------------------------------- - Any suggestions? Lenir

1 0

[Users] problem to detect a loop - load_contacts()/next_contacts()
by Matteo Piazza 08 Dec '05

08 Dec '05

I have realized a serial forking with load_contacts() and next_contacts(), but i have this problem: When the first time arrive the invite messages the contacts are loaded by the function load_contacts(). So when the call flow enters in the failure_route I can load the second contact. At this point the script reload another time the contacts. But in this way Is is impossible to load the third contact, because the script load the preference of the new contact. For example: I call 309(a)domain.it, the script loads 309 contacts and call the fisrt: 309(a)domain.it 309 not answered, the scrip enters in failure route, it loads the 400(a)domain.it and call 400, At this moment I loads the contacts of 400(a)domain.it WRONG!!!!! In which way it is possibile to load the contacts only the firt time. I hope that the problem is clear. Best Matteo ======================================= This is my location table ============================================================ username | domain | contact | q 400 | | sip:400@192.168.9.193 | 1.00 401 | | sip:400@192.168.9.194 | 1.00 308 | | sip:308@192.168.9.181:5060 | 1.00 309 | | sip:309@192.168.9.131;transport=udp | 1.00 309 | | sip:400@192.168.9.97 | 0.80 309 | | sip:401@192.168.9.97 | 0.60 .................... .................... route{ ...... ...... ...... if(load_contacts()){ xlog("L_ERR","LOAD CONTACTS!\n"); }; if(next_contacts()){ xlog("L_ERR","NEXT CONTACT!\n"); }; t_on_failure("1"); t_relay(); exit; } route[1] { log(1,"ROUTE 1 !\n"); if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP log(1,"ROUTE 2 !\n"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { log(1,"FAILURE ROUTE !\n"); if (t_check_status("408")){ if (next_contacts()) { log(1,"NUOVO CONTATTO !\n"); route(2); exit; }; }; exit; } This is the output: 7(5504) LOAD CONTACTS! 7(5504) NEXT CONTACT! 14(5518) FAILURE ROUTE ! 14(5518) NUOVO CONTATTO ! 14(5518) ROUTE 2 ! 6(5502) LOAD CONTACTS! 6(5502) NEXT CONTACT!

2 1

Re: [Users] What's causing this error...
by Bogdan-Andrei Iancu 08 Dec '05

08 Dec '05

Hi, I looked at : http://openser.org/pub/openser/latest/src/openser-1.0.0_src.tar.gz and the table versions look ok. are you using different tables than "location" or "aliases" as parameter of save() or lookup() ? regards, bogdan Wilko van der Ploeg wrote: >I use the openser-1.0.0.src.tar as downloaded on 1 nov 2005 > > >Wilko van der Ploeg > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: donderdag 8 december 2005 13:43 >To: Wilko van der Ploeg >Cc: users(a)openser.org >Subject: Re: [Users] What's causing this error... > >Hi Wilko, > >what version are you using and what;s the date of your checkout? > >regards, >bogdan > >Wilko van der Ploeg wrote: > > > >>After a clean installation of openser (so no 'old tables') a >>openser.cfg script is causing the following error messages : >> >>Dec 8 14:09:17 sip1 /sbin/openser[2419]: register_udomain(): Invalid >>table version (use ser_mysql.sh reinstall) >> >>Dec 8 14:09:17 sip1 /sbin/openser[2419]: domain_fixup(): Error while >>registering domain >> >>Dec 8 14:09:17 sip1 /sbin/openser[2419]: ERROR: fix_expr : fix_actions >> >> > > > >>error >> >>It looks like there is some mismatch in table version in the >>installtion tar.... Does anyone know what should be altered to make it >> >> > > > >>work ? >> >>Wilko >> >>----------------------------------------------------------------------- >> >> >- > > >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > >

1 0

[Users] Openser as client
by Elton Machado 08 Dec '05

08 Dec '05

Hi, Is anyone aware if openser can act like a client and register itself as a UAC in another UA? Regards,

2 5

Re;Re [Serusers] QoS and call quality
by Ashutosh kumar 08 Dec '05

08 Dec '05

Hi, Has anyone tried the QoS tricks given at http://www.voip-info.org/wiki/view/QoS+Linux+with+HFSC, <http://www.voip-info.org/wiki/view/QoS+Linux+with+HFSC,%0bhttp:/www.voip-in fo.org/wiki/view/QoS+Linux,http:/www.voip-info.org/wiki/view/QoS+with+Linux+ using+PRIO+and+HTB> http://www.voip-info.org/wiki/view/QoS+Linux, http://www.voip-info.org/wiki/view/QoS+with+Linux+using+PRIO+and+HTB <http://www.voip-info.org/wiki/view/QoS+Linux+with+HFSC,%0bhttp:/www.voip-in fo.org/wiki/view/QoS+Linux,http:/www.voip-info.org/wiki/view/QoS+with+Linux+ using+PRIO+and+HTB> Do they work? Or has anyone has experience using GlobalIPSound's products to enhance sound quality? Regards, Ashutosh -----Original Message----- From: sip [mailto:sip@arcdiv.com] Sent: Wednesday, December 07, 2005 7:11 PM To: Ashutosh kumar; serusers(a)lists.iptel.org Subject: Re: [Serusers] QoS and call quality SER has no control over the RTP voice stream in ANY way. That's handled by the UAs and your network equipment. N. On Wed, 7 Dec 2005 19:07:37 +0530, Ashutosh kumar wrote > Hi, > Does SER have any control or any can it be configured (if yes, please guide) to implement various QoS mechanisms like Diffserv,RSVP, Admission control etc.., to improve the voice quality and delays and/or jitter? Because lately, we are facing a decrease in call quality in our ser calls. > Or do we have to do this only at the network level, eg. Routers etc..? > > Any poiners/guides will be most welcome. > Regards, > Ashutosh Kumar >

1 0

[Serusers] aliases table
by Xavier TRENTIN 08 Dec '05

08 Dec '05

Hi all, I'm running ser with dbtext, and I can't install mysql on my system. Can anybody tell me if the aliases table is loaded by ser when it starts? I also need to know what exactly each field must contain, so if somebody can give me an example of a line of the aliases table, it will be great. Thanks in advance. B.R. Xavier

2 2

[Users] What's causing this error...
by Wilko van der Ploeg 08 Dec '05

08 Dec '05

After a clean installation of openser (so no 'old tables') a openser.cfg script is causing the following error messages : Dec 8 14:09:17 sip1 /sbin/openser[2419]: register_udomain(): Invalid table version (use ser_mysql.sh reinstall) Dec 8 14:09:17 sip1 /sbin/openser[2419]: domain_fixup(): Error while registering domain Dec 8 14:09:17 sip1 /sbin/openser[2419]: ERROR: fix_expr : fix_actions error It looks like there is some mismatch in table version in the installtion tar.... Does anyone know what should be altered to make it work ? Wilko

2 1

[Serusers] Call forwarding to Cisco PSTN GW and correct charging
by Kyriakos Mavromichalis 08 Dec '05

08 Dec '05

Has anyone implemented call forwarding from SER to a PSTN CISCO gw? Lets say we have the situation below: A is caller from PSTN network B is a sip user that forwarded his number to C C is a pstn number If A calls B then he should be redirected to C , but A should be charged for calling to B , and B should be charged for calling to C. After some digging in the list and Cisco documents found out about CC-Diversion header and remote party id. What i want to do is to have a cdr created by the cisco machine that fills up the calling number field with A number and the redirecting number field with B number. Which header is the one to use ? thanks -- Kyriakos Mavromichalis Otenet Telecom

2 3

[Serusers] Parse Vie Error : REGISTER
by Ricardo Martinez 08 Dec '05

08 Dec '05

Hello list. I updated my SER version to the last Stable version from CVS. I used almost the same ser.cfg file, now, for certain users trying to register in my SER box i'm getting this : REGISTER sip:sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 200.6.65.82:58038;branch=z9hG4bK-245f6be;. From: 5565625856537 <sip:5565625856537@sip.mydomain.com>;tag=5369ee326df7cf6fo1. To: 5565625856537 <sip:5565625856537@sip.mydomain.com>. Call-ID: b3ea4367-c371f5b2(a)10.0.5.120. CSeq: 62661 REGISTER. Max-Forwards: 70. Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c92a5c19d60a1bd1a 9d35e91f35dd97f493",uri="sip:sip.mydomain.com",algorithm=MD5,response="6e894 b948364091bbc2e6bfeeb4e412c",qop=auth,nc=0000107c,cnonce="28e5d319". Contact: 5565625856537 <sip:5565625856537@200.6.65.82:30221>;expires=600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Linksys/RT31P2-3.1.3(LI). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. . Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via: invalid via - end of header in state 6 Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via on: <SIP/2.0/UDP 200.6.65.82:58038;branch=z9hG4bK-245f6be;^M From: 5565625856537 <sip:5565625856537@sip.mydomain.com>;tag=5369ee326df7cf6fo1^M To: 5565625856537 <sip:5565625856537@sip.mydomain.com>^M Call-ID: b3ea4367-c371f5b2(a)10.0.5.120^M CSeq: 62661 REGISTER^M Max-Forwards: 70^M Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c92a5c19d60a1bd1a 9d35e91f35dd97f493",uri="sip:sip.mydomain.com",algorithm=MD5,response="6e894 b948364091bbc2e6bfeeb4e412c",qop=auth,nc=0000107c,cnonce="28e5d319"^M Contact: 5565625856537 <sip:5565625856537@200.6.65.82:30221>;expires=600^M Warning: 399 spa "STUN Server Not Reachable"^M User-Agent: Linksys/RT31P2-3.1.3(LI)^M Content-Length: 0^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M ^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP 200.6.65.82:58038;branch=z9hG4bK-245f6be;^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: get_hdr_field: bad via Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_msg: message=<REGISTER sip:sip.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 200.6.65.82:58038;branch=z9hG4bK-245f6be;^M From: 5565625856537 <sip:5565625856537@sip.mydomain.com>;tag=5369ee326df7cf6fo1^M To: 5565625856537 <sip:5565625856537@sip.mydomain.com>^M Call-ID: b3ea4367-c371f5b2(a)10.0.5.120^M CSeq: 62661 REGISTER^M Max-Forwards: 70^M Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c92a5c19d60a1bd1a 9d35e91f35dd97f493",uri="sip:sip.mydomain.com",algorithm=MD5,response="6e894 b948364091bbc2e6bfeeb4e412c",qop=auth,nc=0000107c,cnonce="28e5d319"^M Contact: 5565625856537 <sip:5565625856537@200.6.65.82:30221>;expires=600^M Warning: 399 spa "STUN Server Not Reachable"^M User-Agent: Linksys/RT31P2-3.1.3(LI)^M Content-Length: 0^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M ^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: receive_msg: parse_msg failed Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via: invalid via - end of header in state 6 Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via on: <SIP/2.0/UDP 200.6.65.82:30219;branch=z9hG4bK-803aca07;^M From: 5565625848753 <sip:5565625848753@sip.mydomain.com>;tag=96e51eca631897b7o0^M To: 5565625848753 <sip:5565625848753@sip.mydomain.com>^M Call-ID: 79a49a8f-c7348aa(a)10.0.5.120^M CSeq: 62704 REGISTER^M Max-Forwards: 70^M Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c968045009a67b029 9bbd27e7a11961b95d",uri="sip:sip.mydomain.com",algorithm=MD5,response="05ba5 577a2171f34a306a21cdb253ac7",qop=auth,nc=0000107a,cnonce="2faea8f6"^M Contact: 5565625848753 <sip:5565625848753@200.6.65.82:30220>;expires=600^M Warning: 399 spa "STUN Server Not Reachable"^M User-Agent: Linksys/RT31P2-3.1.3(LI)^M Content-Length: 0^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M ^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP 200.6.65.82:30219;branch=z9hG4bK-803aca07;^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: get_hdr_field: bad via Dec 7 16:24:58 sipserver ser[2506]: ERROR: parse_msg: message=<REGISTER sip:sip.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 200.6.65.82:30219;branch=z9hG4bK-803aca07;^M From: 5565625848753 <sip:5565625848753@sip.mydomain.com>;tag=96e51eca631897b7o0^M To: 5565625848753 <sip:5565625848753@sip.mydomain.com>^M Call-ID: 79a49a8f-c7348aa(a)10.0.5.120^M CSeq: 62704 REGISTER^M Max-Forwards: 70^M Authorization: Digest username="MT_1911111",realm="sip.mydomain.com",nonce="43930c968045009a67b029 9bbd27e7a11961b95d",uri="sip:sip.mydomain.com",algorithm=MD5,response="05ba5 577a2171f34a306a21cdb253ac7",qop=auth,nc=0000107a,cnonce="2faea8f6"^M Contact: 5565625848753 <sip:5565625848753@200.6.65.82:30220>;expires=600^M Warning: 399 spa "STUN Server Not Reachable"^M User-Agent: Linksys/RT31P2-3.1.3(LI)^M Content-Length: 0^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M ^M > Dec 7 16:24:58 sipserver ser[2506]: ERROR: receive_msg: parse_msg failed Can someone help me here? What does it means? Thanks in advance Regards Ricardo Martinez

2 1

[Serusers] Contract work needed for custom SER configuation
by Andy Pyles 07 Dec '05

07 Dec '05

Please contact me privately if you are very familiar with NATHELPER and AVPOPS modules. I'm looking for a custom configuration solution, that needs to be delivered very quickly. Thanks, Andy

1 0

Re: [Serusers] set timer for "Request Timeout"
by Matteo Piazza 07 Dec '05

07 Dec '05

Hi Jens Can you send me your complete ser.cfg file. I'am interested to see you complete configuration Regards Matteo Hello all, I have following records in the table 'usr_preferences': username attribute value 1000 fwdbusy sip:1001 at example.de 1000 fwdnoanswer sip:1001 at example.de I use these records to get the functionallity of "Call Forwarding - Busy" and "Call Forwarding - No Answer" into the OpenSer - server. I use a failure_route to act on the SIP - messages "Request Timeout (408)" and "Busy(486)" and forward the INVITE - message to the URI saved in the table 'usr_preferences'. See my snippet of the openser.cfg: ----snip --- failure_route[1] { if (isflagset(27) && t_check_error(408)) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); resetflag(27); route(6); break; }; }; if (isflagset(26) && t_check_error(486)) { if (avp_pushto("$ruri", "s:fwdbusy")) { avp_delete("s:fwdbusy"); resetflag(26); route(6); break; }; }; }; ----snap --- These configuration works fine. Is there a possibillity to set a timer for the phone after the server receives the message "Request Timeout" from the phone? Or can I let the server ring for example five times and generate the message? Greetings Jens ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 1

[Serusers] How to Minimize rtp proxy usage for all cases including when behind the same NAT
by S G 07 Dec '05

07 Dec '05

Hello, I have SER setup to only use the rtpproxy if the client is behind a symmetric NAT. This is accomplished by using STUN since STUN will not modify the contact headers if it detects a symmetric NAT. So SER see's the local address in the contact header and only forces rtpproxy for those types of call. When users are behind the same NAT then ser detects this using AVP's and force_rtpproxy is not used for the call. This only works when STUN is not enabled. Since when STUN succeeds the orignal local address of the phone is changed to the public address in the sip message. So when two STUN enabled clients try to call each other from behind the same nat the call fails. The call fails because the NAT does not support harpin of media as most don't. The only way this call would work is if the contacts could be changed back to use their local addresses pre-STUN. How can SER be instructed to use the local address for calls behind the same NAT when STUN is enabled? One way I can think of doing this is to change nathelper to extract the original IP from the VIA header and rewrite the SDP. In the SIP trace i see only IP's in contact headers and SDP are changed when STUN succeeds. Is there a simpler way to accomplish this? Thanks, Sumeet _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

3 5

[Users] CALL FORWARD / PSTN GW
by Kjell Arild Frigstad 07 Dec '05

07 Dec '05

Hello! I am trying to implement Call forwarding. Server runs openser 1.0.0 and I have a Cisco PSTN GW where only defined calling numbers are permitted (because of accounting on the "upstream" provider). My problem is when a call from PSTN is beeing forwarded and I have to "turn it back" to the PSTN gatway. Calls from : 1) Sip server ---> SIP-CALL-FW ---> PSTN GW OK, the calling number is also changed to the number which has turned call forwarding on. The display on the phone also showing the number which has turned Call forwarding on.. 2) PSTN ---> SIP-CALL-FW ---> SIP SERVER Works OK 3) PSTN ----> SIP-CALL-FW ---> PSTN is not working. Dialpeer matching in Cisco not OK because calling number not OK. I can define a way so that all calls can go through the GW. Then I sometimes get a "482 loop detected". The disply on the phone showing the orginal calling number (not the number wich hat turnd Call forwarding on). Is this the right way to "turn" a call back to where it came from? Have a missed a "rebuild" of the message? if (avp_db_load("$ruri","$fw_to") & method=="INVITE") { # CALL Forward log(1, "INFO: AVP_db_load fw_to true \n"); #save called number avp_write("$ruri/username","s:org_to"); #Insert call forward number avp_pushto("$ruri/username","$fw_to"); #is call forward number local or to PSTN if (does_uri_exist()) { log(1, "INFO:Call FW to Local users\n"); if (!lookup("location")) { log(1, "INFO: call FW to local user, not online\n"); route(4); return; }; log(1, "INFO:Call FW to local users online\n"); uac_replace_from("$avp(s:org_to)","sip:$avp(s:org_to)@212.125.213.151:5060"); if (!t_relay()) { sl_reply_error(); return; }; return; } else { log(1, "INFO: Call FW to PSTN\n"); #here is somting wrong. What??? #move Called number as From number and send to PSTN GW uac_replace_from("$avp(s:org_to)","sip:$avp(s:org_to)@212.125.213.151:5060"); rewritehostport("80.239.101.34:5060"); append_hf("P-hint: call fwd to PSTN\r\n"); append_branch(); record_route(); if (!t_relay()) { sl_reply_error(); return; }; }; return; } Cisco trace : calling from 38092382 (PSTN) to 38707130 (on the SIP server) witch is forwarded to 95066833 (PSTN) Call comming in from PSTN to SIP-server : Sent: (from Cisco) INVITE sip:38707130@212.125.213.151:5060 SIP/2.0 Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972 From: <sip:38092382@80.239.101.34>;tag=2A699224-CC6 To: <sip:38707130@212.125.213.151> Date: Fri, 02 Dec 2005 18:41:38 GMT Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60(a)80.239.101.34 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 457212883-1654264282-2767847443-3293903248 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: <sip:38092382@80.239.101.34>;party=calling;screen=yes;privacy=off Timestamp: 1133548898 Contact: <sip:38092382@80.239.101.34:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 253 v=0 o=CiscoSystemsSIP-GW-UserAgent 6274 5968 IN IP4 80.239.101.34 s=SIP Call c=IN IP4 80.239.101.34 t=0 0 m=audio 16788 RTP/AVP 18 8 0 c=IN IP4 80.239.101.34 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 *Dec 2 18:41:38.278: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972 From: <sip:38092382@80.239.101.34>;tag=2A699224-CC6 To: <sip:38707130@212.125.213.151> Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60(a)80.239.101.34 CSeq: 101 INVITE Server: OpenSer (1.0.0 (i386/linux)) Content-Length: 0 Warning: 392 212.125.213.151:5060 "Noisy feedback tells: pid=29232 req_src_ip=80.239.101.34 req_src_port=55717 in_uri=sip:38707130@212.125.213.151:5060 out_uri=sip:95066833@80.239.101.34:5060 via_cnt==1" *Dec 2 18:41:38.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:95066833@80.239.101.34:5060 SIP/2.0 Record-Route: <sip:212.125.213.151;ftag=2A699224-CC6;lr=on> Via: SIP/2.0/UDP 212.125.213.151;branch=z9hG4bK1e4.11a1a9d3.0 Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972 From: <sip:38707130@212.125.213.151:5060>;tag=2A699224-CC6 To: <sip:38707130@212.125.213.151> Date: Fri, 02 Dec 2005 18:41:38 GMT Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60(a)80.239.101.34 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 457212883-1654264282-2767847443-3293903248 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 69 Remote-Party-ID: <sip:38092382@80.239.101.34>;party=calling;screen=yes;privacy=off Timestamp: 1133548898 Contact: <sip:38092382@80.239.101.34:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 253 P-hint: call fwd to PSTN v=0 o=CiscoSystemsSIP-GW-UserAgent 6274 5968 IN IP4 80.239.101.34 s=SIP Call c=IN IP4 80.239.101.34 t=0 0 m=audio 16788 RTP/AVP 18 8 0 c=IN IP4 80.239.101.34 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 Dial peer maching (call forward from SIP server). Calling number is wrong, should be 38707130 (number who has put forwarding on) Calling Number=38092382, Called Number=95066833, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Dec 2 19:44:08.330: //-1/D6732AD9A532/DPM/dpAssociateIncomingPeerCore: Result=NO_MATCH(-1) After All Match Rules Attempt *Dec 2 19:44:08.334: //-1/D6732AD9A532/DPM/dpMatchPeersCore: Calling Number=, Called Number=95066833, Peer Info Type=DIALPEER_INFO_SPEECH *Dec 2 19:44:08.334: //-1/D6732AD9A532/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=95066833 Best regards Kjell Arild

2 2

[Users] Limit simultaneous calls
by Christophe Irles 07 Dec '05

07 Dec '05

Hi, Is there a way to limit the number of simultaneous calls ? According the provider, the bandwith will not permit a lot of simultaneous call so in order to prevent very bad feeling I prefer to limit the use. In a second step, I wondering if this type of restriction can be extended according the phone domain. For example, simultaneous call betwwen two sub-sites of the same enterprise is limited to 10 but all others a limited to 5 for example. I know the ACC module but it's not the one fot this even if the information stored could resolve this point. Perhaps another module already exist ? Thanks, Christophe

6 5

[Serusers] Extracting a substring from r-uri
by Steve Blair 07 Dec '05

07 Dec '05

I would like to extract a substring from the r-uri but cannot see a way to accomplish this task. Is there a function already available? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

2 1

[Users] how to configure SIP OpenSer redirect server
by Ajay Srivastava 07 Dec '05

07 Dec '05

Hi, I made changes in openser.cfg like this to act it as a SIP redirect server: route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); return; }; # rewrite current URI, which is always part of destination ser rewriteuri("sip:parallel@siphub.net <mailto:parallel@siphub.net> :9"); # append one more URI to the destination ser append_branch("sip:redirect@siphub.net <mailto:redirect@siphub.net> :9"); # redirect now sl_send_reply("300", "Redirect"); } then restarted openser, but still not getting redirect messages, should I need to do something more for enabling this option? Please help me and tell me that how can I run my redirect server for SIP. Regards, ajay

2 1

RE: [Users] t_replicate after failure
by Douglas Garstang 07 Dec '05

07 Dec '05

Ok, I worked out that you need to use append_branch, like this: append_branch (proxy1) append_branch (proxy1) t_replicate (proxy3) That's what the devevelopers say anyway... However, all the t_replicate's are going to exactly the same proxy. Here's a post by two people that have reported _exactly_ the same problem. Is there a solution? http://www.archivum.info/serusers@iptel.org/2004-12/msg00268.html Thanks, Doug. -----Original Message----- From: Douglas Garstang Sent: Wednesday, November 30, 2005 4:11 PM To: users(a)openser.org Subject: [Users] t_replicate after failure In the scenario below, if the first t_replicate fails because 192.168.10.5 returns "Not Found", the second t_replicate command does not execute. Why? How can I make it execute? Call me crazy, but this doesn't seem at all intuitive to me. In a functional programming world, if the first fails, the next should still run. if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voip.com", "subscriber")) { www_challenge("voip.com", "0"); exit; }; save("location"); t_replicate("192.168.10.5","5060"); t_replicate("192.168.10.7","5060"); exit; }; Thanks, Doug. _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

3 2

[Users] Monitoring on failure
by Jose Maria Gonzalez 07 Dec '05

07 Dec '05

I want to implement a monitor system when there is a failure route. At first I think to send an snmp message when there was a failure route, but the snmp module is frozen, isn't it? Now I want to write a line in a special log file. So I want to have the normal log file for monitor information (call stablish, call released, etc..) and another log file for errors. I though in use the exec module to inser the errors in the error-log file in this way: exec_dset("echo Error_failure_route >> file_error.log"); Does anyone have a better idea to do this? Because I think that the exec module is slower than use xlog. It is possible to execute that command in background? That openser does not stop the execution until the command has finished. Thanks -- Salu2: *************************************** José María González ***************************************

2 1

[Users] local mysql access problem cont'd
by rick＠eltopia.com 07 Dec '05

07 Dec '05

Alright.. here is some dbg level 9 output of my problem. Basically openser starts, says it starts successfully, and then quits. I think it may have something to do with the fact that it is trying to access a local database even though it has been configured to access a remote one. I am quite sure that it is accessing the proper cfg, cause when I put garbage in it I get an error. Any help, clues, snide remarks, cryptic-guru-riddle-answers would be appreciated. Dec 1 11:48:41 amnesiac openser: 0(0) db_init: Connection 'mysql://openser:openserrw@localhost /openser' not found in pool Dec 1 11:48:41 amnesiac openser: 0(0) new_connection: Opening MySQL connection: mysql://opense r:openserrw@localhost/openser Dec 1 11:48:41 amnesiac openser: 0(0) new_connection: Can't connect to local MySQL server thro ugh socket '/var/lib/mysql/mysql.sock' (2) Dec 1 11:48:41 amnesiac openser: 0(0) register_udomain(): Can not open database connection Dec 1 11:48:41 amnesiac openser: 0(0) domain_fixup(): Error while registering domain Dec 1 11:48:41 amnesiac openser: ERROR: error -1 while trying to fix configuration Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : start Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: unlink_timer_lists : emptying DELETE list Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : emptying hash table Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : releasing timers Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : removing semaphores Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : destroying tmcb lists Dec 1 11:48:41 amnesiac openser: 0(0) DEBUG: tm_shutdown : done Dec 1 11:48:41 amnesiac openser: 0(3791) shm_mem_destroy Dec 1 11:48:41 amnesiac openser: 0(3791) destroying the shared memory lock Dec 1 11:48:41 amnesiac openser: openser startup succeeded my config file is simply the sample config, modified with the suggestions to access a remote mysql database, it looks like this: # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # Uncomment this if you want to use SQL database loadmodule "//lib/openser/modules/mysql.so" loadmodule "//lib/openser/modules/sl.so" loadmodule "//lib/openser/modules/tm.so" loadmodule "//lib/openser/modules/rr.so" loadmodule "//lib/openser/modules/maxfwd.so" loadmodule "//lib/openser/modules/usrloc.so" loadmodule "//lib/openser/modules/registrar.so" loadmodule "//lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//lib/openser/modules/auth.so" loadmodule "//lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- modparam("auth_db", "db_url", "mysql://serusr:testpass@192.168.0.54/ser") # -- 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{ # 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 (!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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} 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; #}; 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); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

2 1

[Serusers] different db for different sip domain
by Iqbal 07 Dec '05

07 Dec '05

Hi As the subject says is it possible to configure ser to auth from different db, based on sip domain And has anyone set this up Iqbal

2 4

[Serusers] How to handle retrasmissions?
by Federico Giannici 07 Dec '05

07 Dec '05

What is the correct way to handle message retrasmissions? Usually t_relay() seems to hadle them correctly. But if a transactions was previously created by some module function, I found that t_relay() (and subsequent sl_reply_error()) generates the following errors: Dec 5 18:49:25 eowyn OpenSER[5654]: ERROR: t_newtran: transaction already in process 0x502a56f8 Dec 5 18:49:25 eowyn OpenSER[5654]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) So, I have the following questions: 1) What is the correct script code to relay a message, even with retrasmissions and transaction already created? 2) Is it OK to simply discard retrasmissions (not forwarding or replying them)? 3) How can I recognize retrasmissions? Is t_lookup_request() the correct function for a test? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 4

Re: [Devel] [Users] nathelper/rtpproxy when both SIP UA are behind same NAT
by Tavis P 07 Dec '05

07 Dec '05

Opps i made a mistake, instead of "avp_write" you need to use "avp_printf" to store the pseudo variable into an AVP and when you are doing the comparison with avp_check you need an alias defined for the second AVP otherwise you might end up comparing the AVP with a literal value modparam( "avpops", "avp_aliases", "dest_ip=i:401" ) avp_printf("i:400", "$si"); avp_printf("i:401", "$rd"); if ( avp_check("i:400", "eq/$dest_ip/g") ) { xlog("L_INFO", "\"$si\" == \"$rd\" "); } Norman Brandinger wrote: > I agree that the solution that was presented is BAD for the reasons > stated by both of us and probably more :) > > Perhaps using $Ri after the successful lookup() and comparing it to > the $src_ip AVP would work. > > The problem was trying to compare a pseudo variable with an AVP. > > > The doc <snipped below> does not state that pseudo variables can be > used by avp_write(). > > Pseudo-variables can be used with following modules of OpenSER: > > * avpops - function “avp_printf()” > * xlog - functions “xlog()” and “xdbg()” > > I did try to use avp_write() to save a pseudo variable to an AVP but > was unsuccessful. > > Do you have an example of writing a pseudo variable to an AVP ? > > Regards, > Norm > > > > Tavis P wrote: > >> Another, more lightweight solution, is: >> >> After a successful lookup(location) you can compare the received IP >> address ($si) with the destination domain/ip address returned from the >> "lookup" function call ($rd), i think these two pieces of information >> are reliable for this purpose >> >> This way you don't have to make extra database calls to determine >> locality >> >> Also if you want to compare pseudo variables you can first write them to >> an AVP using "avp_write" and then you can make use of "avp_check" with >> data sourced from a pseudo variable >> >> Norman Brandinger wrote: >> >> >> >>> I have looked into the problem related to saving saving resources by >>> not calling rtpproxy or mediaproxy when both the caller and callee are >>> behind the same NAT. >>> >>> This topic has been discussed many times however, there have been no >>> "working" examples, or even partial examples posted (that I'm aware >>> of). >>> >>> Below is a solution to the problem, but is a BAD hack at best. >>> >>> This solution is BAD because AVP's are added to the usr_preferences >>> table during the registration process, but there is no notification >>> that a registration has expired and is no longer valid. A cron job >>> could be run periodically to delete all received_ip AVP's with no >>> matching entry in the location table, but this is another hack to fix >>> the first hack. >>> >>> The solution is BAD because a database calls must be made to save the >>> received_ip address into an AVP, then another database call must be >>> made to reload the value just so that it can be tested against the >>> caller's $src_ip. A pseudo variable $Ri already exists that contains >>> the value we use in received_ip, but I have not been able to find a >>> way for avp_check() to use $Ri. >>> >>> The solution is BAD because it will fail when the SIP device(s) are >>> behind more than a single NAT router. I believe this is not an issue >>> for 95% of the users and 99% of the small office or home users. >>> >>> I think that a better solution would be to enhance avp_check() to >>> accept pseudo variables for the "value" parameter. This would remove >>> the need for any database calls. >>> >>> I'm cross-posting this response to the developers list to bring this >>> to their attention and ask for suggestions (it's possible that I've >>> totally missed something). If a decision is made to open the avpops >>> module to pseudo variables (other than just avp_print()), I would >>> suggest the following be looked at: >>> >>> avp_check() the value parm should accept pseudo variables >>> avp_write() the value parm should accept pseudo variables >>> avp_pushto() the name parameter should accept pseudo variables >>> avp_op() the value parameter should accept pseudo variables >>> >>> Regards, >>> Norm >>> >>> >>> 1) During REGISTER processing, place the following code. I would >>> suggest that you put the code "after" all authentication checks and >>> "after" the save("location") statement. >>> >>> # Delete any previously saved IP addressess from the user. >>> avp_db_delete("$from/username","s:received_ip"); >>> >>> # Save the source IP address of the user into an AVP called >>> received_ip. >>> # The saved IP address should be the public address of the NAT router. >>> avp_write("$src_ip", "s:received_ip"); >>> >>> # Save the AVP received_ip into the usr_preferences tables associated >>> with the user that just registered >>> avp_db_store("$from/username","s:received_ip"); >>> >>> 2) During INVITE processing (or wherever you make the decision to use >>> or not use rtpproxy / mediaproxy) , place the following code. >>> >>> >>> #--------------------------------------------------------------------------- >>> >>> >>> # Are the caller and callee behind the same NAT ? >>> >>> #--------------------------------------------------------------------------- >>> >>> >>> avp_db_load("$to/username", "s:received_ip"); >>> if (avp_check("s:received_ip", "eq/$src_ip")) { >>> setflag(CALLER_AND_CALLEE_ ARE_BEHIND_THE_SAME_NAT); >>> }; >>> >>> >>> >>> _______________________________________________ >>> Devel mailing list >>> Devel(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/devel >>> >>> >>> >> >> >> >> >> > > >

3 9

[Serusers] QoS and call quality
by Ashutosh kumar 07 Dec '05

07 Dec '05

Hi, Does SER have any control or any can it be configured (if yes, please guide) to implement various QoS mechanisms like Diffserv,RSVP, Admission control etc.., to improve the voice quality and delays and/or jitter? Because lately, we are facing a decrease in call quality in our ser calls. Or do we have to do this only at the network level, eg. Routers etc..? Any poiners/guides will be most welcome. Regards, Ashutosh Kumar

2 1

[Serusers] avp invite timer not working?
by Kyriakos Mavromichalis 07 Dec '05

07 Dec '05

Im trying to set the invite timer to 6 seconds when i have a call to pstn gw but it does not seem to affect anything, This is how i set the timer: avp_write("5", "inv_timeout"); any ideas? -- Kyriakos Mavromichalis Otenet Telecom

3 7

[Users] Re: [Devel] unixODBC module compilation error
by Bogdan-Andrei Iancu 07 Dec '05

07 Dec '05

Hi Lenir, the warning is fixed on cvs - please update and retry. thanks for report, bogdan Lenir wrote: > Guys, > > > > I downloaded the most recent version of CVS, I get the following > compilation error, when compiling unixodbc module: > > make[1]: Entering directory `/usr/src/sip-server/modules/unixodbc' > > Compiling list.c > > gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > -minline-all-stringops -malign-double -falign-loops -mtune=athlon > -Werror -DNAME='"openser"' -DVERSION='"1.1.0-dev3 "' > -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -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 list.c -o list.o > > list.c: In function `insert': > > list.c:56: warning: control reaches end of non-void function > > make[1]: *** [list.o] Error 1 > > make[1]: Leaving directory `/usr/src/sip-server/modules/unixodbc' > > > > Any ideas? > > > > Lenir > > > >------------------------------------------------------------------------ > >_______________________________________________ >Devel mailing list >Devel(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/devel > >

1 0

[Users] UAC to BroadVoice
by Scott MacGregor 07 Dec '05

07 Dec '05

I am trying to route to a BroadVoice PSTN gateway using the UAC. In the failure_route the uac_auth command fails with the following errors: ERROR:uac:parse_authenticate_body: parse error in <DIGEST realm="BroadWorks",algorithm=MD5,nonce="1133904594381"> around 0 ERROR:uac:uac_auth: failed to parse auth hdr body code: failure_route[3] { if( t_check_status("401|407") ) { if( uac_auth() ) { append_branch(); route(1); } } } Thanks,

2 1

[Serusers] How to register MSN Messenger4.6 to non-default port of ser
by Manuel Seleiro 07 Dec '05

07 Dec '05

Manuel Seleiro Portugal!!!

2 2

[Users] symmetric fw
by Paul van Schagen 06 Dec '05

06 Dec '05

Hi, I am using STUN and that works well most of the time. I do have a problem with UA's who sit behind a symmetric firewall. ( a Microsoft server with ISA 2004 to be exact ). I understand this is a well known problem. What practical solutions can I use in combination with openser ? Thanks for any help Best regards Paul van Schagen

2 1

[Users] unixODBC module compilation error
by Lenir 06 Dec '05

06 Dec '05

Guys, I downloaded the most recent version of CVS, I get the following compilation error, when compiling unixodbc module: make[1]: Entering directory `/usr/src/sip-server/modules/unixodbc' Compiling list.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -Werror -DNAME='"openser"' -DVERSION='"1.1.0-dev3 "' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -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 list.c -o list.o list.c: In function `insert': list.c:56: warning: control reaches end of non-void function make[1]: *** [list.o] Error 1 make[1]: Leaving directory `/usr/src/sip-server/modules/unixodbc' Any ideas? Lenir

1 0

[Serusers] hierarchical VoIP system
by Joao Pereira 06 Dec '05

06 Dec '05

Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt

3 4

[Serusers] Writing TCL CLient to populate SER
by Selvasundar 06 Dec '05

06 Dec '05

hi, i have installed the SER in a REd hat linux machine. i want to write a test stub in TCL , and the stub will throw SIP messages to SER and will receive the same from SER. any one , please guide me in this regard. Thanks in advance Rgds, John 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 confidential or privileged information. If you are not the intended recipient, please notify the sender or administrator(a)tataelxsi.co.in

1 0

[Users] session termination
by JF 06 Dec '05

06 Dec '05

Hi, I've been playing around with TM callbacks and the TM API trying to build a module which mantains dialog info in order to permit session termination. For now I managed (in a very experimental way) to have a function exposed to the script which initiates (appropriately built with dialog info) BYE transactions to UAs and effectively terminates the call. Later this could be done via FIFO or unix socket command taking some kind of "dialog identifier" parameter... But sending BYEs to terminate calls only works if the dialog is already in the CONFIRMED state. For early dialogs, CANCELs should be used. But in the TM API there is no cancel function. Is there any other way of canceling a request on behalf of an UA from within an (open)ser module? Or would the TM API need to be extended? Thanks in advance for any insights. JF

1 0

[Users] openserctl
by Douglas Garstang 06 Dec '05

06 Dec '05

How can we change the database, database user name and database password that the openserctl utility tries to connect with? You can set these in openser.cfg with modparam, but if you can't also set them for openserctl, what's the point? Thanks. Douglas Garstang

2 3

[Serusers] Call hunting w/lcr, how to set diff q values?
by Brian McCrary 06 Dec '05

06 Dec '05

Hello, I'm trying to setup a hunt group using the lcr module and different q values. When I try to add a user location with a different q value using the FIFO commands, the only q values that are accepted is 1.00 or lower, such as 0.99. However, these all seem to qualify as the same value as all of my locations are still contacted at the same time. Does anyone have this working, and how did you add the user locations in regards to the q value? Thanks, Brian

2 4

[Serusers] Web registration problem through ser_fifo
by Andrey Kouprianov 06 Dec '05

06 Dec '05

What is wrong with my PHP code, I have no idea. Could you please check it and let me know what's wrong. Im using SER 0.9.3, but ser_fifo documentation is not available, so i used one from ser 0.8.12 (admin guide). Im sure there's a little error, but I can figure out which one. Btw, below the code is the output from message log <? /* config values */ $domain="sip.mydomain.org"; $user="webme"; $web_aor="sip:".$user."@".$domain; $expires=3600; $type="student"; $fifo="/tmp/ser_fifo"; /* open reply fifo */ $replyfilename="webfifo_".rand(); $replypath="/tmp/".$replyfilename; echo "Initiating your request...<p>"; /* open fifo now */ $fifo_handle=fopen( $fifo, "w" ); if (!$fifo_handle) { exit ("Sorry -- cannot open fifo: ".$fifo); } /* construct FIFO command */ $fifo_cmd=":ul_add:".$replyfilename."\n". "location\n". //table $web_aor."\n". //aor "sip:".$user."@".$_SERVER['REMOTE_ADDR']."\n". //contact $expires."\n". //expires "0.00\n". //priority "0\n\n"; /* create fifo for replies */ system("mkfifo -m 666 ".$replypath ); /* write fifo command */ if (fwrite( $fifo_handle, $fifo_cmd)==-1) { unlink($replypath); fclose($fifo_handle); exit("Sorry -- fifo writing error"); } fclose($fifo_handle); /* read output now */ if (readfile( $replypath )==-1) { unlink($replypath); exit("Sorry -- fifo reading error"); } unlink($replypath); echo "<p>You are now signed in<p>"; ?> ///MESSAGE LOG Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: read_line: request line too long Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: ul_add: flags expected Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: fifo_server: command must begin with :: location Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: fifo_server: command must begin with :: sip:webme@203.159.32.50;type=student Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: fifo_server: command must begin with :: 0.00 Dec 6 16:35:54 sip /usr/local/sbin/ser[206]: ERROR: ul_add: flags expected

2 1

Re: [Users] errors when making a call!
by Klaus Darilion 06 Dec '05

06 Dec '05

messa(a)innsof.com wrote: > Hi Klaus > > this command does not work for me! > but, my openser works with those errors! You have to install the "ngrep" application. E.g. "apt-get install ngrep" if you are using debian based distros. klaus > > bests regards! > Serge > > > > >>looks like a bad formated SIP message. >> >>Post the SIP signaling dump (ngrep -t -W byline port 5060) >> >>klaus >> >>messa(a)innsof.com wrote: >> >>>Hi all >>> >>> When imake calls, signalisation goes well but, when, i hang up my >>>client, i have no voice in any client! >>>And my openser server display methese errors: >>> >>>0(4335) lookup(): Error while extracting address of record >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> >>> 0(4335) WARNING: do_action:error in expression >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) extract_aor(): Error while parsing Address of Record >>> 0(4335) lookup(): Error while extracting address of record >>> 0(4335) Warning: sl_send_reply: I won't send a reply for ACK!! >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> >>> 0(4335) loose_route: Error while parsing Request URI >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> >>> 0(4335) WARNING: do_action:error in expression >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) extract_aor(): Error while parsing Address of Record >>> 0(4335) lookup(): Error while extracting address of record >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> >>> 0(4335) WARNING: do_action:error in expression >>> 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) >>> 0(4335) extract_aor(): Error while parsing Address of Record >>> 0(4335) lookup(): Error while extracting address of record >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy >>> >>>How can i fix those errors? >>> Thanks for help >>> Serge >>> >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > > >

1 0

[Serusers] from_uri in the LCR table
by Eric Haskins 06 Dec '05

06 Dec '05

Can Anyone tell me is it possible to route calls based on from_uri with the LCR Module?? I see the from_uri field but I have tried using %(a)192.168.1.1 & 192.168.1.1 I replaced 192 address with an actual IP of another SER box that is Trusted. No matter what I put in there it still routes to the % only group. Even when I dial the prefix related to that entry. Eric

1 1

[Users] transfer not possible!!
by Jayesh Nambiar 06 Dec '05

06 Dec '05

hi everyone, I was just trying to transfer the call to another user. All users are registered at the same openser server. Now when I receive a call, is it not possible to transfer the call to another IP user. I thought REFER mehtod does this, but i think I am wrong. Is this possible, can somebody pls clear my understanding regarding this concept. thanks in advance!! jayesh --------------------------------- Enjoy this Diwali with Y! India Click here

1 0

[Serusers] UNIX socket and FIFO concurrently?
by Andy Pyles 06 Dec '05

06 Dec '05

HI, is it possible to have a unix socket and FIFO at the same time? Is it the same API for both? Andy

2 3

[Serusers] SER and Music on Hold
by Waldo Rubinstein 05 Dec '05

05 Dec '05

I'm new to SER and was wondering how are people offering music on hold with it. Is it possible? Is the solution simply to send the call to Asterisk? What about SEMS? Does it offer music on hold? Thanks, Waldo

2 2

Re: [Serusers] nathelper/rtpproxy when both SIP UA are behind same NAT
by Jan Henkins 05 Dec '05

05 Dec '05

Thanks for the info! :-) If I do not succeed in getting around this hairpinning issue, I'll have to replace my Netgear gateway with one of these. On Mon, 2005-12-05 at 12:29 -0500, sip wrote: > On Mon, 05 Dec 2005 11:59:18 +0200, Jan Henkins wrote > > Hello all, > > > > On Sun, 2005-12-04 at 18:13 -0500, sip wrote: > > > > > If the gateway has no issues with hairpinning, then yes, it's quite possible. > > > I use a Linksys gateway at home and a slightly older, pre-sip-proxy version > > > of Astaro linux firewall at work, and we have multiple UAs behind each in the > > > NAT space of our firewall. They can call each other. They can all outside. All > > > based off registrations with a SER server on the outside of the network. > > > > Could you perhaps tell me the model number of your Linksys gateway? > > > > I'm using a Linksys WRT54G (not sure which series). It's got the latest > Linksys firmware on it (if that makes any difference). > > N. -- Regards, Jan Henkins

1 0

[Serusers] Blind transfer and accounting
by Bjorn Asmul 05 Dec '05

05 Dec '05

Has anyone successfully dealt with "Blind Transfer", such that the original party will be billed for the call he transfers? -- Bjorn

1 0

[Users] Serial forking Help!!
by Matteo Piazza 05 Dec '05

05 Dec '05

I become crazy to realize a serial forking. I try to use load_contacts() and next_contscts(). This is my openser.cfg nd my location table: In teory when I call 309(a)192.168.9.98 the location with the q value equal to 0.80 it is the first but it is processed only the location 309(a)192.168.9.131. Help!!! # # $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:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-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/avpops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/lcr.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("registrar", "append_branches", 1) #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","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("auth_db","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("lcr","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("avpops","avp_url","mysql://openser:heslo@127.0.0.1/openser") modparam("avpops","avp_table","usr_preferences") modparam("tm", "fr_timer", 8) modparam("tm", "fr_inv_timer", 10) modparam("tm", "wt_timer", 5) #modparam("avpops","avp_aliases","fwdbusy=i:665") # ------------------------- 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; }; if (method =="INVITE" && uri =~"^sip:0[0-9]*@*"){ log(1, "Check 1 Start PSTN Call\n"); rewritehostport("192.168.9.97:5061"); t_relay(); exit; } if (method == "CANCEL"){ t_relay(); } if (method =="BYE"){ t_relay(); } # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} 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("create-net.it", "subscriber")) { www_challenge("create-net.it", "0"); exit; }; 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"); }; if(load_contacts()){ xlog("L_ERR","LOAD CONTACTS!\n"); } xlog("L_ERR","CONTACT <$ru> !\n"); t_on_failure("1"); t_relay(); #route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { if (next_contacts()) { xlog("L_ERR","NEXT CONTACT <$ru> !\n"); append_branch(); t_relay(); } else { xlog("L_ERR","ERROR!\n"); } } This is my location table ============================================================ username | domain | contact | q 400 | | sip:400@192.168.9.193 | 1.00 308 | | sip:308@192.168.9.181:5060 | 1.00 309 | | sip:309@192.168.9.131;transport=udp | 1.00 309 | | sip:400@192.168.9.97 | 0.80 -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 3

[Users] SER and OpenBSD/amd64 3.8
by Federico Giannici 05 Dec '05

05 Dec '05

Yesterday I upgraded the version of the operating system of our sip server from OpenBSD amd64 3.7 to 3.8. Now I noticed that domain name lookups seems to don't work anymore and the INVITES are replyed with 478. Here is an example of the logs: Dec 5 13:11:03 eowyn OpenSER[15740]: ERROR: mk_proxy: could not resolve hostname: "voip.eutelia.it" Dec 5 13:11:03 eowyn OpenSER[15740]: ERROR: uri2proxy: bad host name in URI <sip:800900800@voip.eutelia.it:5060;user=phone> Dec 5 13:11:03 eowyn OpenSER[15740]: ERROR:tm:t_forward_nonack: failure to add branches Obviously "voip.eutelia.it" is resolved by dig on the same machine. Anybody had the same problem? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

5 7

RE: [Serusers] SER + Radius
by Ricardo Martinez 05 Dec '05

05 Dec '05

Hi again That's the way SER send the information to the RADIUS server, using Digest attributes. Regarding to the "not readable outputs" i don't know why that happen to you, in my case i use RADIATOR and i can read the Digest attributes without any problem. What RADIUS server are you using? Ricardo.- -----Mensaje original----- De: Sait Karalar [mailto:saitkaralar@gmail.com] Enviado el: Lunes, 05 de Diciembre de 2005 11:06 Para: Ricardo Martinez Asunto: Re: [Serusers] SER + Radius Hi, I read that tutorial, but when I send REGISTER request to the radius, I got some un-readable chars only. I expect SIP-Auth-Name or SER-Auth-Name, got instead Digest-Auth-Name etc... Digest-XXXX 's. and values are really not readable outputs. just some chars. so, I send this mail to get more information. :) thanks. On 12/5/05, Ricardo Martinez < rmartinez(a)redvoiss.net <mailto:rmartinez@redvoiss.net> > wrote: Hello There is a RADIUS HOW TO somwhere in the website www.iptel.org <http://www.iptel.org/> , you can authenticate SIP users via RADIUS, and do accounting also. About managing routes, maybe you can try LCR module, but it works only through a mysql local database, perhaps you can try do somethign with the AVP_RADIUS module... Good Luck Ricardo.- -----Mensaje original----- De: Sait Karalar [mailto: <mailto:saitkaralar@gmail.com> saitkaralar(a)gmail.com] Enviado el: Sábado, 03 de Diciembre de 2005 8:59 Para: serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> Asunto: [Serusers] SER + Radius Hi list, I am so new to list SER. I just tried to I want to authenticate SIP users via RADIUS and want accounting with radius. also if possible, ask route to radius also. but I did not manage it ;( any simple sample? I read documents etc. no good sample found. -- gMail : google mail Try it! It is the best http://www.gmail.com <http://www.gmail.com/> -- gMail : google mail Try it! It is the best http://www.gmail.com <http://www.gmail.com>

1 0

[Serusers] SER + freeRADIUS with mysql integration
by Noel Sharpe 05 Dec '05

05 Dec '05

Hi List I want to start using RADIUS authentication, and I was wondering whether anyone has integrated the SER database (schema) with freeRADIUS. I have a web application that is using data from the the SER database, and I would if possible rather not change the web application. Has anyone here got an sql.conf file with queries adjusted for the SER schema that they are willing to share? Kind Regards Noel

1 0

[Users] unixodbc module
by Bogdan-Andrei Iancu 05 Dec '05

05 Dec '05

Hi everybody, thanks to Marco Lorrai - marco.lorrai(a)abbeynet.it - contribution, openser has new DB support module. I find it very useful as, as its own, it's a middle layer to other DB drivers For info, see http://www.unixodbc.org/ and for supported drivers see http://www.unixodbc.org/drivers.html. Marco will maintain the module - I already grant you CVS access on the project. Please feel free to work on it. The online documentation is available at: http://www.openser.org/docs/modules/1.1.x/unixodbc.html Marco, thank again! regards, bogdan

3 3

RE: [Serusers] LCR script?
by Ricardo Martinez 05 Dec '05

05 Dec '05

Hello Kyriakos. Check this thread: http://lists.iptel.org/pipermail/serusers/2005-July/022461.html Good Luck Ricardo Martinez.- > -----Mensaje original----- > De: Kyriakos Mavromichalis [mailto:kyriakos@otenettel.com] > Enviado el: Lunes, 05 de Diciembre de 2005 6:56 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] LCR script? > > > Hi, > has anyone written a script for creating the sql tables > needed for the lcr > module? > > > -- > Kyriakos Mavromichalis > Otenet Telecom > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Serusers] Question regarding UDP TCP translation
by Ken Stauffer 05 Dec '05

05 Dec '05

Did you get this working? I am trying to do the same thing. Thanks

1 0

[Serusers] 483-too many hops
by crazy 05 Dec '05

05 Dec '05

Hi all Today I found a error when I call other one by Voip phone, indicates that "483-too many hops", but I use the X-lite in computer without this error and everything is OK! What can I do for it ? in hurry!!!!!

4 3

RE: [Serusers] SER + Radius
by Ricardo Martinez 05 Dec '05

05 Dec '05

Hello There is a RADIUS HOW TO somwhere in the website www.iptel.org <http://www.iptel.org/> , you can authenticate SIP users via RADIUS, and do accounting also. About managing routes, maybe you can try LCR module, but it works only through a mysql local database, perhaps you can try do somethign with the AVP_RADIUS module... Good Luck Ricardo.- -----Mensaje original----- De: Sait Karalar [mailto:saitkaralar@gmail.com] Enviado el: Sábado, 03 de Diciembre de 2005 8:59 Para: serusers(a)lists.iptel.org Asunto: [Serusers] SER + Radius Hi list, I am so new to list SER. I just tried to I want to authenticate SIP users via RADIUS and want accounting with radius. also if possible, ask route to radius also. but I did not manage it ;( any simple sample? I read documents etc. no good sample found. -- gMail : google mail Try it! It is the best http://www.gmail.com <http://www.gmail.com>

1 0

[Serusers] T1 timer?
by Andreas Granig 05 Dec '05

05 Dec '05

Hi, I'm currently facing a situation where UAC A sends an INVITE to UAC B via SER, and SER sends two completely identical INVITEs to UAC B, the second about 50ms after the first one (no second INVITE from UAC A though). Could this be a retransmit? Seems to be a little bit too early (should be 500ms according to the rfc), and I don't think it's a parallel fork because the requests are identical regarding dst-ip/port and the message itself, and there is only one usrloc record. Any other possibilities? And another question: does SER itself retransmit requests when relaying, or is this the responsibility of the calling UAC? If so, how is the T1 timer defined in SER? I only found this one: ./modules/tm/config.h:#define RETR_T1 1 So is T1 defined as one second? Thanks, Andy

2 2

[Users] errors when making a call!
by messa＠innsof.com 05 Dec '05

05 Dec '05

Hi all When imake calls, signalisation goes well but, when, i hang up my client, i have no voice in any client! And my openser server display methese errors: 0(4335) lookup(): Error while extracting address of record 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> 0(4335) WARNING: do_action:error in expression 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) extract_aor(): Error while parsing Address of Record 0(4335) lookup(): Error while extracting address of record 0(4335) Warning: sl_send_reply: I won't send a reply for ACK!! 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> 0(4335) loose_route: Error while parsing Request URI 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> 0(4335) WARNING: do_action:error in expression 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) extract_aor(): Error while parsing Address of Record 0(4335) lookup(): Error while extracting address of record 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) ERROR: parse_sip_msg_uri: bad uri <sip:> 0(4335) WARNING: do_action:error in expression 0(4335) ERROR: parse_uri: uri too short: <sip:> (4) 0(4335) extract_aor(): Error while parsing Address of Record 0(4335) lookup(): Error while extracting address of record 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy 0(4335) force_rtp_proxy2: incorrect port in reply from rtp proxy How can i fix those errors? Thanks for help Serge

2 1

[Serusers] Authenticate to Active Directory
by Bob McCormick 05 Dec '05

05 Dec '05

Does anyone know if there is a way to authenticate SER to MS Active Directory? I'd like to use an existing user database instead of having to manage users myself, and most of our users are in Active Directory. Has anyone tried this? What are most SER users using for authentication? Is anyone here tying into any kind of existing organization user database?

4 3

[Serusers] LCR script?
by Kyriakos Mavromichalis 05 Dec '05

05 Dec '05

Hi, has anyone written a script for creating the sql tables needed for the lcr module? -- Kyriakos Mavromichalis Otenet Telecom

1 0

[Serusers] Communication between SER as Proxy and a SIP Server
by Anthony Chapellier 05 Dec '05

05 Dec '05

Hi, I need to configure SER as a SIP proxy and that all SIP requests go through SER. To do it, you normally add SER as an outbound proxy into a routing table in your SIP server. But actually I can't modify the server... So, the solution I wish to implement is a way to register SER in SIP server as an "UAC" when it receives a REGISTER request from a real one. So first things first, is it possible ? I think you could do it by doing the following : - Save UAC real location. - Rewrite Contact Header Field in SIP packet request with SER IP adress and forward this request to SIP server. Then when you receive a request from SIP server, SER searches for UAC real location and forward request to it. So finally, how could you do it ? I tought I could use subst() function but it seems I can't master this function perfectly to make it work. So does anyone have an idea ? Thanks, Anthony

1 0

[Serusers] nathelper/rtpproxy when both SIP UA are behind same NAT
by Rafael R. GV 05 Dec '05

05 Dec '05

Hi I need to let the UAs speak directly to each other without force rtpproxy, Is there any solution for this issue? rafael ---------- Forwarded message ---------- From: Rafael J. Risco G.V. <rafael.risco(a)gmail.com> Date: Apr 22, 2005 2:17 PM Subject: UA´s behind same NAT using na thelper/rtpproxy, No Audio... To: serusers(a)lists.iptel.org I am using avpops to enable SER to see if the UAs are located behind the same address and now I need to let the UAs speak directly to each other (without force rtpproxy), please someone see my configuration below and send some advice. rafael ----. if (!lookup("location")) { log(1,"unable to locate user \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"); . . # -------------- Same NAT Call Routing (no force rtpproxy) ---- route[3]{ log(1," route[3]: UAs are in the same nat, NO force_rtp_proxy "); ### Do something!!!! if(!t_relay()){ sl_reply_error(); }; }

7 10

[Serusers] Which all methods does SER supports????/
by nitin ganjsinghani 05 Dec '05

05 Dec '05

Hi all can anyb plz tell me that which all methods r supported by SER regards Nitin

1 0

[Serusers] SER + AD
by Andrej Sirk 05 Dec '05

05 Dec '05

Hello, somebody knows how to integrate SER with AD. Maybe with LDAP module? thanks

1 0

[Serusers] Replicate REGISTER from SER to non-SER sip server
by Matt morris 05 Dec '05

05 Dec '05

Hi all, I have this scenario: A SIP Client needs to register with both a SER server and a non-SER sip server (different domains). How can I achieve this in SER's config? I have seen this example in the list to forward the REGISTER message to other SER servers: if(method == "REGISTER") { if(src_ip==<IP of peer1> || src_ip==<IP of peer2>) { save_noreply("location"); } else { if(!www_authorize("<my domain>", "subscriber")) { www_challenge("<my domain>", "0"); break; } if(!check_to()) { sl_send_reply("403", "Use To=username next time"); break; } if(!save("location")) { sl_reply_error(); break; } forward_tcp("<IP of peer1", 5060); forward_tcp("<IP of peer2", 5060); } break; } Would it work for my scenario? I tried it but it seems the forward_tcp line gave me this error: "forward_req: ERROR: cannot forward to af 2, proto 2 no coresponding listening socket" I'm using SER 0.8.14, thanks all in advance. _________________________________________________________________ Powerful Parental Controls Let your child discover the best the Internet has to offer. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

1 0

[Serusers] Multiple Invites
by ser list 05 Dec '05

05 Dec '05

Hi, I am new to Ser, and hence I need some help on the following: 1. Someone makes a PSTN call and my cisco router forwards the call to my SER sip server. By default, my router will forward it to an account on SER, such as 1000. 2. Upon receiving the request, SER should make ALL registered phones in the same domain ring. When someone picks up the phone, other phones should stop ringing. I am able to implement point 1, but how can I make all phones ring at the same instance? I am thinking of doing something like the following for INVITES : if(uri =~"sip:1002@serproxy.com"){ rewriteuri("sip:1003@serproxy.com"); forward(uri:host, uri:port); rewriteuri("sip:1004@serproxy.com"); forward(uri:host, uri:port); } Can someone pls advise me on the best approach? Regards, zan

1 0

[Serusers] ser.cfg for proxy and registrar
by Simon HEBBO 04 Dec '05

04 Dec '05

Hi all, I am trying to install SER with mysql on a debian distribution. The installation went very good. But I am not able to configure it to work as registrar and proxy in the same time. Ser will serve a private network without any connection to the internet because i need to test some security issues on SIP. I think with a good ser.cfg file it will works. So if anyone have a ser.cfg file that enable ser to work as a proxy and registrar in the same time on a private lan, and could send it to me i will be very thankfull. Thank you all, Simon HEBBO

4 3

Re: [Serusers] UAs and SER behind NAT
by Ladislav Andel 04 Dec '05

04 Dec '05

Actaully, at the end I changed it in the source code nathelper.c and recompiled it and it works fine for me. I will look at your solution too.. Thank you, Lada Michal Matyska wrote: > I have done a little test and have found, that we have to update the doc; > the description of the tests is in opposite way... > - nat_uac_test("1") tests for private IP in contact header > - nat_uac_test("2") tests for mismatch of via IP/port and real IP/port > > Sorry for confusing. If you are getting true for the nated clients even for > nat_uac_test("2"), try to put force_rport() into your script before the > nat_uac_test... > > Michal > > On Sun, Dec 04, 2005 at 10:15:01AM +0100, Ladislav Andel wrote: > >> Thank you for your reply but nat_uac_test("1") will test src_ip >> against RFC1918 addresses anyway. >> I have tried and even seen it in nathelper.c . >> >> Ladislav >> >> > > >

1 0

[Users] Features support cfg file ?
by Norman Brandinger 04 Dec '05

04 Dec '05

Hi Paul, I've placed an example implementation online at the following location: http://openser.org/dokuwiki/doku.php?id=avp_examples Hope it helps. Regards, Norm

3 3

[Serusers] Announcement: New FTP Server with SIP related software
by Jan Janak 04 Dec '05

04 Dec '05

Hello, We have a new FTP server containing the collection of your favorite SIP and RTP related software. The FTP server is reachable either using FTP at ftp://ftp.iptel.org (use anonymous username and your email address as password), or through the web interface at http://ftp.iptel.org ftp.iptel.org is a public FTP server dedicated to freely available SIP (Session Initiation Protocol, RFC3261) based software. Here you can find everything that you will need to implement robust and highly scalable SIP based server systems. Among others, this archive contains: * SIP Express Router (SER) * RTP proxy server for NAT traversal (RTPProxy) * Web based administration interface (SERWeb) * SIMPLE based instant messaging and presence server with support for XCAP * voicemail and announcement server (SEMS) We also keep a local copy of IETF RFCs and Internet drafts. Should you have any problems, requests, or suggestions for improvements, do not hesitate to contact us by e-mail at <ftpmaster(a)iptel.org>. The contents of the FTP server will be updated and expanded regularly, it is not limited to iptel.org or SER related software. Jan.

3 4

[Serusers] UAs and SER behind NAT
by Ladislav Andel 04 Dec '05

04 Dec '05

Hello, I'm sorry for this basic question, but I have SER running on my LAN with (RFC1918) addresses with UAs on the same network. Then there are UAs registering from behind NAT to this SER on the LAN network. (SER is not listening on any public IP and it's just used in the private network) I came across this issue. If a UA on the LAN network is registering to SER the logic of ser.cfg recognize the UA is behind NAT after nat_uac_test. If I want to introduce NAT traversal with RTPproxy for this private network nat_uac_test will always return that the UA is behind NAT because it has RFC1918 IP address. The RTPproxy will be used at all times for UAs on the network or even I introduce STUN for NATed UAs I know this is not a common usage of SER because it is invented to be used on the global internet but I was just wondering is there any possibility not to use RTPproxy at all times for UAs on the private network? Maybe this is a silly question but it would help me for my testing purposes with SER where I will not have to use my setup with usage of public IPs. Thank you for any thoughts. Ladislav

3 2

[Users] Creating permenant contacts with FIFO ul_add
by Doug Meredith 04 Dec '05

04 Dec '05

I am trying to use FIFO to manage permanent contacts. I can create contacts now, but can't figure out what value to specify for the expiry. A positive value represents a real expiry in seconds. Negative numbers, "Never", and "Permanent" all result in "400 Invalid expires format". A zero results in "500 Error while adding contact". The information at http://www.openser.org/dokuwiki/doku.php?id=fifo_relay#ul_add is more of an example. It doesn't really explain the arguments. Does anyone know how to specify that a contact is permanent? The command "openserctl ul add" is able to create permanent contacts, so it must be possible. Doug -- Doug Meredith (doug.meredith(a)systemguard.com) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com

2 2

[Users] FIFO response "400 ul_add: replicate expected"
by Doug Meredith 04 Dec '05

04 Dec '05

I am using OpenSER 1.0. When I send the following to the FIFO: ----------------- :ul_add:ans location 81(a)dev.skyridge.com sip:34@sirloin.skyridge.com 3600 1 ----------------------- I get the response "400 ul_add: replicate expected". I have done a google search without finding anything helpful. Can anyone explain what this means? Doug -- Doug Meredith (doug.meredith(a)systemguard.com) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com

2 2

[Users] dispatcher relaying instead of dipatching?
by Rick Richardson 03 Dec '05

03 Dec '05

On my test network I have set up a dispatcher server that points to two proxy servers that hit an external database. I have a simple route script on the dispatcher that should be routing requests to the two proxy servers. dispatcher .51 caller .99 recipient .100 proxy1 .52 proxy2 .53 I'm making call requests from .99 to .100, with .51 used as the proxy server. The problem I'm getting, is that the INVITE is being sent directly to the recipient without passing through the proxys, the dispatcher is passing the message directly. At least i think that is what is happening. Here is the sip INVITE: INVITE sip:rick@192.168.0.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.99:5060;rport;branch=z9hG4bK837723126 From: <sip:rick@192.168.0.51>;tag=679508646 To: <sip:rick@192.168.0.100> Call-ID: 125992616(a)192.168.0.99 CSeq: 20 INVITE Contact: <sip:rick@192.168.0.99:5060> Max-Forwards: 5 User-Agent: Linphone-1.0.1/eXosip Subject: Phone call Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 352 So it seems to me that the invite request is being sent to .51 and it is forwarding directly to .100 without dispatching the message to .52 or .53 here is my dispatcher config: # $Id: dispatcher.cfg,v 1.1.1.1 2005/06/13 16:47:37 bogdan_iancu Exp $ # sample config file for dispatcher module # debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 fifo="/tmp/openser_fifo" # for more info: sip_router -h # ------------------ module loading ---------------------------------- loadmodule "//lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("2", "2"); forward(uri:host,uri:port); # t_relay(); }

1 0

[Serusers] Failure route weirdness....
by sip 03 Dec '05

03 Dec '05

Okay... I'm a little stumped on what's going on here, so I'll start with a general explanation before I bother posting my (likely hard to read) ser.cfg. I'm having a weird issue when calling from X-Lite to a SNOM 320 phone (I haven't tried the SNOM 190 with a different firmware yet, but I don't see that it would make much difference). I have a failure route which handles voicemail and such, and I have the invite timer set to 27 seconds (roughly 4.5 standard US rings). Now, when I call from X-Lite to X-Lite, or from Leadtek 8051 to X-Lite, or from SNOM to X-lite, or from SNOM to SNOM, or from Leadtek to Leadtek, etc, etc... everything works EXACTLY as it's supposed to. I have log messages telling me when the failure route is being entered, and it hits it at right on 27 seconds. However, when I call TO the SNOM phone from any other device besides another SNOM phone, I never hit the failure route until a 408 timeout at 58 seconds. Now, I can't think of a reason for a UA to overwrite the invite timer on SER or even how such overall weirdness would occur, but it's 58 seconds every time like clockwork.. error 408. At that point, it enters the failure route and does its processing (forwarding the call to voicemail as far as SER is concerned but not as far as the UA is concerned... which just receives an error message). Any ideas? Also... when I dial SNOM to SNOM, the SIP method I get in the ACC is PRACK (no INVITE). Can someone fill me in on what a PRACK is? Is there something I should be looking at to figure out why calling one UA behaves differently than other UAs? N.

2 2

[Serusers] SER + Radius
by Sait Karalar 03 Dec '05

03 Dec '05

Hi list, I am so new to list SER. I just tried to I want to authenticate SIP users via RADIUS and want accounting with radius. also if possible, ask route to radius also. but I did not manage it ;( any simple sample? I read documents etc. no good sample found. -- gMail : google mail Try it! It is the best http://www.gmail.com

1 0

[Users] Features support cfg file ?
by Paul van Schagen 03 Dec '05

03 Dec '05

Hello, I have setup openser and I have a request to incorporate an unconditional forward feature which can be turned on by dialing a "*21+ number" and can be turned of by dialing #21#. Can anyone provide me with an example piece of .cfg file that can do this ? best regards Paul

1 0

[Users] Handeling Diversion - expired?
by Helge Waastad 03 Dec '05

03 Dec '05

Hi, I have a question more in a general aspect. Since the draft-levy-sip-diversion-08 has expired, should a Proxy strip such headers? According to RFC3261,the UAS MUST ignore headers-fields not defined in the specification or any supported extension... I've realized that some systems need Diversion headers for voicemail (I know Asterisk doesn't, but Cisco Unity does) Any comments? -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 1

[Serusers] STUN support
by Jan Henkins 02 Dec '05

02 Dec '05

Hello all, Is there anything that needs to be done to the ser cfg file to support STUN? If so, could you please share that config snippet? Thanks in advance. -- Regards, Jan Henkins Cell: +27-84-951-4334

2 2

[Serusers] NAT transversal
by Voipers Portugal 02 Dec '05

02 Dec '05

---------- Forwarded message ---------- From: Voipers Portugal <voipers(a)gmail.com> Date: Dec 2, 2005 9:08 AM Subject: Re: [Serusers] NAT transversal To: jan(a)henkins.za.net I think you should start by reading the nathelper module. There you'll find the information that you will need. If you want, try the OnSip config file for NAT, it's quite self explaning, and it works. Jose Simoes On 12/2/05, Jan Henkins <jan(a)henkins.za.net> wrote: > > Hi, > > I'm trying to set up ser (version 0.9.4) on a server that is not behind > any form of NAT. I need to get to this ser server from behind NAT > firewalls (ADSL). To test I'm using a variety of software (linphone, > Twinkle etc.), and I also have a Grandstream Budgetone 100 that I'm > using. I tested the Grandstream phone successfully on my FWD account > with STUN, however try as I might I can not log on to my server. What is > the best way to accommodate this type of setup? At the moment I need a > simple nudge into a direction so that I can research it properly. > > Thanks in advance! > > -- > Regards, > Jan Henkins > Cell: +27-84-951-4334 > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Users] DB Integration
by Douglas Garstang 02 Dec '05

02 Dec '05

All, Thinking of modifying the OpenSER scripts/mysqldb.sh script, and adding some new tables. One for example would be table that Asterisk uses to log CDR's for calls. It can use it's own database instances, but it seems to make more sense to put it all into one database. Would also like to add some new columns to existig tables, such as subscriber. Does anyone have any comments as to why this would be a bad idea? I can't think of any reason why it would cause any problems for OpenSER if there where additional fields or tables. Thanks, Doug

2 1

[Serusers] Planning SER + RADIUS + mySQL
by Paco Orozco 02 Dec '05

02 Dec '05

Hello, I'm using SIP in order to be able to call to my PBX extensions. I've configured SER + asterisk and It's working well, but I've got some troubles with authentication: I want to be able to autehnticate with RADIUS in an LDAP (It's working) And I'd like to manage groups with mySQL tables (provided by SER package). But... this is my problem... How can create users in mySQL, It's obligatory to register (via web, for example) to create records is mySQL table? I've visit this homepage: http://www.ethworld.ethz.ch/technologies/sipeth/notes and I've saw some modules very useful, but w/o documentation. Has anyone used it? Thanks in advance -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida.

1 0

[Serusers] SER Stress Test & Sipsak
by Ramin Nikaeen 02 Dec '05

02 Dec '05

Friends, Does anyone know of any other tool for stress-testing the SER other than Sipsak? Is sipsak good?! Thanks ramin

4 3

[Serusers] using subst function from textops module
by Anthony Chapellier 02 Dec '05

02 Dec '05

Hi, I wish to use subst function and looked into module textops documentation to know how to use it. The documentation provides an example which is using a variable or flag named \u. Looks like this '\u' is a replacement for the current uri of the SIP request. # replace the uri in to: with the message uri (just an example) if ( subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/t:\1\u\2/ig') ) {}; So, I would like to know if there are more variables like this one that can be used with subst ? And are there variables too for log function that can be used the same way ? Thanks,

1 0

[Serusers] SER Comment
by Selvasundar 02 Dec '05

02 Dec '05

hi andres , i request your help to install SIP SER as a proxy . THanks selva 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 confidential or privileged information. If you are not the intended recipient, please notify the sender or administrator(a)tataelxsi.co.in

2 1

[Users] Crash (core)
by Lars Sundqvist 02 Dec '05

02 Dec '05

I have switched from SER to OpenSER in our production enviroment since I need a few of the "new" modules, I am however having troubles with OpenSER hanging and crashing, so much so, that I had to change back. There is no clues in the logs to why this is happening but i have a coredump from a crash: (gdb) where #0 0x0052f7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x005707d5 in raise () from /lib/tls/libc.so.6 #2 0x00572149 in abort () from /lib/tls/libc.so.6 #3 0x08054863 in del_lump (msg=0x0, offset=186, len=-135181242, type=HDR_OTHER_T) at data_lump.c:288 #4 0x00f8bd20 in replace_from (msg=0x8130d98, from_dsp=0xbff977a0, from_uri=0x0) at from.c:207 #5 0x00f8e57b in w_replace_from2 (msg=0x8130d98, dsp=0x812f700 "\030W\022\b\v", uri=0x0) at uac.c:311 #6 0x08051703 in do_action (a=0x8126658, msg=0x8130d98) at action.c:677 #7 0x080529e3 in do_action (a=0x8126680, msg=0x8130d98) at action.c:773 #8 0x0805297b in do_action (a=0x8126a58, msg=0x8130d98) at action.c:773 #9 0x0805297b in do_action (a=0x8126f80, msg=0x8130d98) at action.c:773 #10 0x0805297b in do_action (a=0x8127058, msg=0x8130d98) at action.c:773 #11 0x0805297b in do_action (a=0x812b6d0, msg=0x8130d98) at action.c:773 #12 0x080529e3 in do_action (a=0x812b6f8, msg=0x8130d98) at action.c:773 #13 0x080529e3 in do_action (a=0x812b720, msg=0x8130d98) at action.c:773 #14 0x0805297b in do_action (a=0x812b7f8, msg=0x8130d98) at action.c:773 #15 0x08052b42 in run_actions (a=0x8122da0, msg=0x8130d98) at action.c:773 #16 0x080753e9 in receive_msg ( buf=0x80eb300 "INVITE sip:987654321@xxx.xxx.xxx SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bK0ffd9a6274aeb347617736a6da35e918\r\nTo: 987654321<sip:987654321@xxx.xxx.xxx>\r\nFrom: 123456789<si"..., len=943, rcv_info=0xbff98b80) at receive.c:160 #17 0x0808dd1c in udp_rcv_loop () at udp_server.c:464 #18 0x08064e7f in main_loop () at main.c:1033 #19 0x08065615 in main (argc=3, argv=0xbff98d34) at main.c:1570 (Numbers and sip domain edited) Anyone who has a clue whats going wrong? Lars Sundqvist

2 1

[Serusers] Does SER support SIP SUBSCRIBE/NOTIFY methods
by Vivek 02 Dec '05

02 Dec '05

Hi All, Myself Vivek n i m working as a Q A engineer at VeriSign Inc., India. Due to some requirement I am planning use SER but before that i just want to assure whether it supports SUBSCRIBE/NOTIFY and other extended SIP methods. Please help me by providing the correct information. Thanks, Vivek --------------------------------- Yahoo! Shopping Find Great Deals on Gifts at Yahoo! Shopping

2 1

[Serusers] MYSQL
by Colin Jordan 02 Dec '05

02 Dec '05

Every time I activate the command below ser does not start. Please help. fifo_db_url="mysql://ser:heslo@localhost/ser" DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

2 1

[Serusers] SER performance degrades dramatically when invalid users send REGISTER messages continuously
by tulay sazak 02 Dec '05

02 Dec '05

Hello everybody, I am using SER 0.9.3 and freeRADIUS 1.0.4 with digest authentication. I am having problems when users with invalid credentials (wrong username or password) try to register to SER. The same thing happens when I insert some records to usergroup table as Reject type in Radius DB. SER is becoming unfunctional in this case in our system. It cannot perform registration and cannot make call for even valid subscribers. SIP clients with invalid credentials try to register and send REGISTER messages continuously. A SIP client sends 20 request/min and there are 7-8 clients available at the moment. i.e. it is not a big traffic. But every client is tried to be authenticated using digest during registration and at the beginning of the call. In such a case, SER slows down very much. Retransmissions starts and SER becomes unfunctional. Did anyone experienced this problem? Do you have any idea why does it happens? Thanks in advance.. Regards, Tulay Sazak NGN, Borusan Telekom --------------------------------- Yahoo! DSL Something to write home about. Just $16.99/mo. or less

4 3

[Serusers] Multiple PSTN routes
by Kyriakos Mavromichalis 02 Dec '05

02 Dec '05

Hi, i have been doing some searching for how to route to a standby pstn gateway when primary pstn gateway fails. From what i gathered we can either have a failure route that sends the call to the second pstn gateway after an unsuccessful attempt of relaying to the primary gateway, or do a parallel forking to both and the gateway that is active responds back. Which is the best method ? Are there any resources i can read for setting this up? -- Kyriakos Mavromichalis Otenet Telecom

2 2

[Users] mediaproxy problem
by Klaus Darilion 02 Dec '05

02 Dec '05

Hi! I'm having problems with the mediaproxy which I can not resolve. Sometimes, when the session is terminated, mediaproxy does really terminate the session and the ports keep allocated. At the moment I'm having 400 open sessions and mediaproxy uses 80% of the CPU. Attached you can see the logfile of mediaproxy (to avoid line breaks). The first call does not terminate correctly and the ports are still opened. The second call terminates correctly. Can someone help me digging into this problem? regards klaus Dec 2 10:30:49 philipp mediaproxy[440]: command request 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167 1.2.32.167:19514:audio 1.2.32.167 1.2.32.167 remote 3.4.240.143 remote Cisco-SIPGateway/IOS-12.x info=from:06503836629@1.2.32.167,to:43159966509977@1.2.32.160,fromtag:16588354-2D7,totag: Dec 2 10:30:49 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: started. listening on 1.2.32.83:36888 Dec 2 10:30:49 philipp mediaproxy[440]: command execution time: 1.33 ms Dec 2 10:30:57 philipp mediaproxy[440]: command lookup 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167 3.4.240.143:16434:audio 3.4.240.143 1.2.32.167 remote 1.2.32.160 unknown Linksys/PAP2-2.0.13(LSb) info=from:06503836629@1.2.32.167,to:43159966509977@1.2.32.160,fromtag:16588354-2D7,totag:df301db23bf537dbi0 Dec 2 10:30:57 philipp mediaproxy[440]: command execution time: 0.73 ms Dec 2 10:30:57 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: called signed in from 3.4.240.143:16434 (RTP) (will return to 3.4.240.143:16434) Dec 2 10:30:57 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:19514 (RTP) (will return to 1.2.32.167:19514) Dec 2 10:31:01 philipp mediaproxy[440]: session 28667818-624D11DA-A322FFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:19515 (RTCP) (will return to 1.2.32.167:19515) Dec 2 10:33:22 philipp mediaproxy[440]: command delete BF809954A357C8D45FCF51D81299(a)84.151.154.100 info= Dec 2 10:33:22 philipp mediaproxy[440]: command execution time: 0.27 ms Dec 2 10:36:54 philipp mediaproxy[440]: command request 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167 1.2.32.167:18412:audio 1.2.32.167 1.2.32.167 remote 1.2.33.19 remote Cisco-SIPGateway/IOS-12.x info=from:069911160036@1.2.32.167,to:4359966366102@1.2.32.160,fromtag:165E1650-2D0,totag: Dec 2 10:36:54 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: started. listening on 1.2.32.83:36894 Dec 2 10:36:54 philipp mediaproxy[440]: command execution time: 1.52 ms Dec 2 10:36:57 philipp mediaproxy[440]: command lookup 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167 1.2.33.19:32240:audio 1.2.33.19 1.2.32.167 remote 1.2.32.160 unknown CSCO/6 info=from:069911160036@1.2.32.167,to:4359966366102@1.2.32.160,fromtag:165E1650-2D0,totag:000dedfb04cc2d723b9fccb6-64a00b22 Dec 2 10:36:57 philipp mediaproxy[440]: command execution time: 0.60 ms Dec 2 10:36:58 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:18412 (RTP) (will return to 1.2.32.167:18412) Dec 2 10:36:58 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: called signed in from 1.2.33.19:32240 (RTP) (will return to 1.2.33.19:32240) Dec 2 10:37:01 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: caller signed in from 1.2.32.167:18413 (RTCP) (will return to 1.2.32.167:18413) Dec 2 10:37:08 philipp mediaproxy[440]: command delete 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167 info= Dec 2 10:37:08 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: 510/505/1015 packets, 30800/30300/61100 bytes (caller/called/relayed) Dec 2 10:37:08 philipp mediaproxy[440]: session 224B803-624E11DA-A32BFFDE-5D565222(a)1.2.32.167: ended. Dec 2 10:37:08 philipp mediaproxy[440]: command execution time: 0.55 ms

1 0

[Serusers] SER questions
by Andrew Gutierrez 02 Dec '05

02 Dec '05

We are evaluating a hosted VOIP platform from a company called Lignup ( www.lignup.com <http://www.lignup.com/> ). They tell us we need a Session Border Controller to work as an intermidaetary between the customers and the VOIP PBX and the PSTN. They use or recommend NexTone. Will or does SER perform the same functions. Business Class Internet Services Andy Gutierrez Phoenix Internet <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=2922+W.+Clarendon+Ave.&csz=P hoenix%2C+AZ+85017&country=us> 2922 W. Clarendon Ave. Phoenix, AZ 85017 <mailto:andy@phxinternet.net> andy(a)phxinternet.net IM: andyznetandyznet <http://www.phoenixinternet.net> www.phoenixinternet.net tel: fax: 602-234-0917 x 115 602-234-0699 <https://www.plaxo.com/add_me?u=21475002481&v0=312722&k0=883971886> Add me to your address book... <http://www.plaxo.com/signature> Want a signature like this?

3 2

[Users] (open)SER and ICE
by Christophe Irles 02 Dec '05

02 Dec '05

Hi, As I'm a newbie on this subject, I'm wondering how two SIP devices with ICE compatibility will interact with a (open)SER+Mediaproxy implementation since ICE implementation could permit to avoid RTP flow on the (open)SER server. Is there any update to do on the conf file on the (open)SER ? Since they are ICE compatible, the SIP flow (signalization) will be analysed by this two devices in order to create a direct RTP flow ? Or is it because i know on the (open)SER side that this devices are ICE compatible I send a special SIP message ? In fact I want to know if a server update is necessary or if it's "free" to have this great feature. :) Thanks, Christophe

7 7

[Users] a router error
by messa＠innsof.com 02 Dec '05

02 Dec '05

Hi all I use d-link 704P router. When i want to make a call, signalisation is good, but, when the client is hang up, i have voice to the client which is connect in the D-link LAN and the other client has no voice. My server has a public IP. When i'm find the log of the router, i see these messages: unallowed access from 192.168.0.53:to 66.249.93.104:protocol=6 rule=3(by firewall) 192.168.0.53, is the adress of the client behind the d-link lan, the client which has voice and cannot transmit it! what means theses messages? how can i solve the router problem to make my calls? please help best regards! Serge

2 1

[Users] OpenSER attempting local mysql connection
by Rick Richardson 02 Dec '05

02 Dec '05

I have openser.cgf pointing openser to use a remote database using this line: modparam("auth_db", "db_url", "mysql://serusr:mypass@192.168.0.54/ser") Upon initializing the auth_db module it seems to be insisting on connecting locally (which fails) I compiled all the modules for mysql5 using the devel libs from the mysql official redhat enterprise rpms. I can add/rem users to this remote database using openserctl. When I start openser I get this output: 0(3076) WARNING: fix_socket_list: could not rev. resolve 192.168.0.52 0(3076) WARNING: fix_socket_list: could not rev. resolve 192.168.0.52 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.0.52 [192.168.0.52]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.0.52 [192.168.0.52]:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: amnesiac:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: amnesiac:5060 stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) new_connection: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) 0(0) register_udomain(): Can not open database connection 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration any ideas? Rick

2 4

[Serusers] NAT transversal
by Jan Henkins 02 Dec '05

02 Dec '05

Hi, I'm trying to set up ser (version 0.9.4) on a server that is not behind any form of NAT. I need to get to this ser server from behind NAT firewalls (ADSL). To test I'm using a variety of software (linphone, Twinkle etc.), and I also have a Grandstream Budgetone 100 that I'm using. I tested the Grandstream phone successfully on my FWD account with STUN, however try as I might I can not log on to my server. What is the best way to accommodate this type of setup? At the moment I need a simple nudge into a direction so that I can research it properly. Thanks in advance! -- Regards, Jan Henkins Cell: +27-84-951-4334

1 0

[Users] Can anyone share his openser.cfg..in which openser is routing calls to pstn gateway
by Lokesh Kumar 01 Dec '05

01 Dec '05

Hello, I am trying to configure pstn gateway with openser.when i start openser with default openser file it works fine, but i one direction. i can dial on my SIP user agent from outside. but when i am using pstn.cfg file as a openser.cfg(after rename). I am unable to even start openser server. it gives me errors and crashed.I puts my pstn gateway ip's in that file where requested. Please please help Regards

4 6

[Users] AVPs are lost on relayed INVITE errors
by Federico Giannici 01 Dec '05

01 Dec '05

I'm using OpenSER 1.0.0 on OpenBSD 3.7 amd64. I have a strange problem with the accounting: I set a couple of AVPs for every message that arrives at the server. I'm sure they are there because they are written in the syslog logging. Sometimes, when an INVITE is relayed (with transactions) and receives an error (488, 422, etc.), in the SQL logging there is no more presence of the AVPs! Is this a known problem? How can I avoid this? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

4 20

[Users] Re: [Devel] nicer compiling
by Bogdan-Andrei Iancu 01 Dec '05

01 Dec '05

Hi, thanks to Anand, you have now the liberty to get a more friendly output during compilation. Just do: NICER=1 make all normal "make all" still produces same amount of output - it;s the default since it's useful to see the compilation flags...at least for me... also I got rid of the messages regarding the missing dependency files (.d) during first compilation.... so, have a nicer compiling ;) regards, bogdan Anand Kumria wrote: >Hi, > >Attached is a diff that makes it nicer to compile openser. Should you >want to see the exact compilation sequence Q="" make <...> will show it >to you. > >Thanks, >Anand >

2 2

[Users] Serial fork with AVPOPS
by Matteo Piazza 01 Dec '05

01 Dec '05

Hi, I want to realize a serial fork. With the function avp_db_load() I load the addresses to forward the call and the function avp_pushto() rewrites the address. I have this problem (see the debug output): I try to call 309(a)192.168.9.97 309 does not answer Go to failure_route to write 400(a)192.168.9.97 The problem is that it comes executed another time avp_db_load(), and I don't understand why the voicemail is executed two times. Thank's for all! 7(30331) AVP found for r-uri <sip:309@192.168.9.131;transport=udp> for Call forward on Busy 14(30345) AVP inserted: r-uri <sip:400@192.168.9.97> for Call forward on Busy 6(30329) AVP found for r-uri <sip:400@192.168.9.193> for Call forward on Busy 14(30345) R-uri <sip:309@192.168.9.97> for VoiceMail 14(30345) ERROR:tm:t_forward_nonack: no branch for forwarding 14(30345) ERROR: w_t_relay_to: t_relay_to failed 14(30345) R-uri <sip:309@192.168.9.97> for VoiceMail 14(30345) ERROR:tm:t_forward_nonack: no branch for forwarding 14(30345) ERROR: w_t_relay_to: t_relay_to failed 9(30335) AVP found for r-uri <sip:400@192.168.9.193> for Call forward on Busy 9(30335) ERROR: t_newtran: transaction already in process 0xb5da3d88 6(30329) ERROR:tm:t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 9(30335) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) 6(30329) ERROR:tm:t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 route{ .................. if (avp_db_load("$ruri","")) { xlog ("L_ERR", "AVP found for r-uri <$ru> for Call forward on Busy\n"); } t_on_failure("1"); .......... } failure_route[1] { if (avp_pushto("$ruri", "s:fwdbusy")) { xlog ("L_ERR", "AVP inserted: r-uri <$ru> for Call forward on Busy\n"); avp_delete("s:fwdbusy"); append_branch(); t_relay(); t_on_failure("1"); } else{ avp_pushto("$ruri","s:voicemail"); xlog ("L_ERR", "R-uri <$ru> for VoiceMail\n"); rewritehostport("192.168.9.97:5061"); append_branch(); t_relay_to_udp("192.168.9.97","5061"); t_relay(); } ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 4

[Serusers] Creating CDRs from SERs accouning records
by Frank Fischer 01 Dec '05

01 Dec '05

Hi all i'm looking for a free/opensource application that generates CDRs (not XDRs, must not be rated) from the accounting data/events generated by SER (i think this is usually called normalization). I think this would have to be an application that is able to match INVITEs and BYEs and form CDRs with call start timestamp, call duration and so on. I had a look at the list of accounting software on the iptel.org website, but most of there applications are highly integrated accounting/billing solutions, far more than what i need (and also most of them seem to be commercial products). In the mailinglist archive i only found some references to CDRTool which seems also to be commercial. Obv. there must be a free version of CDRTool but neither i'm sure whether it implements exactly what i need so i could "grab it" for my own purposes nor could i find a place to download the free version). Does anyone know if there is such a kind of application available? Or any other idea how to deal with my requirements? Thanks a lot for your help Frank

9 18

[Users] hi all
by hemant kumar 01 Dec '05

01 Dec '05

hello list, i am new to openser. i am trying to setup openser and asterisk together where asterisk will work as a voicemail server. i am trying to find a sample config which can help me understand how to route the calls to asterisk when user is unavailable, busy or time out after sometime. thanks, hemant

2 1

[Serusers] assertion failed in t_retransmit_reply; race condition in t_newtran?
by Valentin Nechayev 01 Dec '05

01 Dec '05

Hi, we had got series of SER cores with failed assertion in t_retransmit_reply(): assert(t->uas.response.dst.send_sock); Essential backtrace part: #0 0x48167fc4 in kill () from /usr/lib/libc.so.4 #1 0x481a993e in abort () from /usr/lib/libc.so.4 #2 0x481858d3 in __assert () from /usr/lib/libc.so.4 #3 0x48281da3 in t_retransmit_reply () from /usr/local/lib/ser/modules/tm.so #4 0x48277433 in t_newtran () from /usr/local/lib/ser/modules/tm.so This place from ser.cfg: if (!t_newtran()) { sl_send_reply("500", "could not create transaction"); break; }; response.dst.send_sock is filled in init_rb(), which is called from t_newtran(). t_newtran() has the following: === cut === UNLOCK_HASH(p_msg->hash_index); /* now, when the transaction state exists, check if there is a meaningful Via and calculate it; better do it now than later: state is established so that subsequent retransmissions will be absorbed and will not possibly block during Via DNS resolution; doing it later would only burn more CPU as if there is an error, we cannot relay later whatever comes out of the the transaction */ if (!init_rb( &T->uas.response, p_msg)) { LOG(L_ERR, "ERROR: t_newtran: unresolvable via1\n"); put_on_wait( T ); t_unref(p_msg); return E_BAD_VIA; } return 1; === end cut === Does there race condition between UNLOCK_HASH(), init_rb() and parallel t_newtran() from another SER process exist? Details: FreeBSD 4.10, SER 0.9.3 (with PortaOne additions, but not essential for the question). -- Valentin Nechayev PortaOne Inc., Software Engineer mailto:netch@portaone.com

1 0

[Users] Serial forking
by Bogdan-Andrei Iancu 01 Dec '05

01 Dec '05

Hi, I just committed serial proper forking support into core - it was migrated from LCR module. I mean proper, since it has q value support and it can be used by any module without any inter-module dependencies. The idea behind was to allow to all module that performs parallel forking to do also serial forking - exec, enum, registrar, etc. There are two new script functions : *serialize_branches(n)* : it inherits the functionality of load_contacts() from LCR; gets all parallel branches and convert them into AVPs for serial forking; numerical parameter 'n' says if any previous AVP should be removed (if non-0) or not (if 0). Returns true is no error (even if no serialization happened). *next_branches()* : it inherits the functionality of next_contacts() from LCR; get (based on q value) the next contact(s) to be used in sequential forking. Returns true only if a new contact was got to be used. The AVP containing the branches is accessible only via alias - its ID is not configurable or visible; the alias (automatically exported by core) is "serial_branch" - it is visible from any module that uses the avp core aliasing system. here are some examples of how to uses this new functionality: 1) enum doing serial forking: { ..... enum_query("e164.arpa.","voice"); serialize_branches(1); if (!next_branches()) { sl_send_reply(...no enum found...); exit; } t_on_failure("1"); t_relay(); } failure_route[1] { if (next_branches()) { t_relay(); } } 2) same for registrar, just replace enum_query() with lookup("location"). NOTE that this will not work yet if there are nated contacts: there is no support to save also the flags and the dst_uri in order to perform proper nat traversal. The code was not yet heavily tested, so any help on this is highly appreciated. On the TO-DO list, related to this subject are: 1) SRV lookup to use this functionality to provide any additional entries. 2) full support for registrar for nat traversal 3) remove load/next_contacts() from LCR (once this new functions prove to be stable). regards, bogdan

4 5

RE: [Users] Sending Authorisation Credentials
by Douglas Garstang 01 Dec '05

01 Dec '05

I don't think this will work. That means not only would the Asterisk box need to let REGISTER requests through without authentication, it would also need to let every other type of request through (INVITES come from OpenSER as well). You can't drill down to that level of detail in Asterisk and tell it to not authenticate for REGISTERs from OpenSER, but to authenticate for INVITE requests. Doug -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, November 30, 2005 2:54 PM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Sending Authorisation Credentials configure asterisk to accept REGISTER without authentication, if it is received from your proxy (IP address) klaus Douglas Garstang wrote: > I have user agents registering with OpenSER. I wanted to 'copy' the registration to an Asterisk server, so I modified a standard openser.cfg as such: > > if (method=="REGISTER") { > > # Uncomment this if you want to use digest authentication > if (!www_authorize("voip.com", "subscriber")) { > www_challenge("voip.com", "0"); > exit; > }; > > save("location"); > t_replicate("192.168.10.7","5060"); # Added this line... > exit; > }; > > I just stuck the t_replicate() call in there. OpenSER sends the REGISTER packet to the Asterisk box who requests credentials. How can I get OpenSER to send credentials (ie act as a UAC) for a given user, to a UAS? > > Thanks, > Doug > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

3 2

[Serusers] Ser + mysql auth
by Paco Orozco 01 Dec '05

01 Dec '05

Hiya, I'm trying to configure ser in order to authenticate to mysql database. I've installed ser from debian package version: ser 0.9.3 (i386/linux) I've look for help into Google but I can't find any valuable help. This is the error message: 0(0) DEBUG: init_mod: auth 0(0) auth module - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/lib/ser/modules/sl.so] 0(0) DEBUG: init_mod: auth_db 0(0) auth_db module - initializing 0(0) find_mod_export: <db_use_table> in module sql not found 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) ERROR: auth_db_bind: unable to bind to the database module 0(0) init_mod(): Error while initializing module auth_db Can anyone help me? -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida.

1 0

[Serusers] Mediaproxy question
by George Lambson 01 Dec '05

01 Dec '05

Anyone know what force_rport() does? or does anyone have additional info on the exported functions for this module?

2 1

[Serusers] SER as a Proxy
by Selvasundar 01 Dec '05

01 Dec '05

hi , this is john, i need to write a TESt Stub in TCL for P/S/I CSCFs , need to install as a Proxy for the test setup. do not find simple guidelines to install the same Thanks & Regards john 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 confidential or privileged information. If you are not the intended recipient, please notify the sender or administrator(a)tataelxsi.co.in

1 0

[Serusers] Adding a user
by Xavier TRENTIN 01 Dec '05

01 Dec '05

Hi all, I use ser with dbtext. I have two questions. 1) I 've added many fields in my subscriber table, and I'd like to know if all the table is loaded in memory when ser starts or if it only loads the fields it needs? 2)When I add users in my subscriber table, I must restart ser to load the new accounts. I thought there were a specific serctl which worked with dbtext, but I can't find it. Can any one give me a link where I can dowload it, and does it allow to insert new users in the text table and in the in loaded table at the same time? If not, how can I add my new users directly in memory? Thanks in advance. B.R. Xavier.

1 0

FW: [Serusers] Repeated Register Message
by Pravesh 01 Dec '05

01 Dec '05

Hi Chan, I also faced a similar problem in which when UA sends REGISTER to SER from different IP address than what is present in the contact header of the REGISTER message, then SER actually stores two AORs for the UA, i.e. one AOR is the actual contact header value & other is the same username with the IP address & port from which I sent the REGISTER message. I am not sure for the reason of this kind of behavior. Anyone from the SER developer list should be able to answer this query. So also forwarding this message to developers list also. Regards, Pravesh -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of tcchan Sent: Wednesday, November 30, 2005 8:57 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Repeated Register Message Hi All, I encountered a situation that somehow a UA keep on Registering with the server with the same IP but on different port. So SER, thought that there are multiple instances of the same username and save the various location into the database. The problem is it registers so often that the memory is full not very long. I suspect it has something to do firewall but I am not sure. Has anyone encounter this problem before? Any hint?? For the time being, is there a way to limit the number of REGISTER message that the SER will take? Regards, TC Chan _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] can't connect to RTP proxy
by lo ™ 01 Dec '05

01 Dec '05

hi, i read the onsip.org <http://onsip.org> getting started manual. i tried again with ser and rtpproxy but i received same errors... then i decided to try with mediaproxy: i downloaded ser-0.9.0 (emerged from gentoo portage) and mediaproxy-1.4.2. i followed this instructions: http://www.voip-info.org/wiki-MediaProxy but i cannot even start it (neither --help works) $ /etc/init.d/mediaproxy start : bad interpreter: No such file or directory and: $ /usr/local/mediaproxy/mediaproxy.py : No such file or directory i got ensured that EOL are not dos-style and i tried that a simple file called test.py (it just prints something) just to make sure that python works... sorry, i understand it is not properly a ser problem but i don't know where else to ask... thank you so much!

2 5

[Serusers] Any mailing list about using SER?
by lin9k＠yahoo.com 01 Dec '05

01 Dec '05

Hi, I have encountered a problem while trying to use Radius with ser. Whenever I try to load the module "auth_radius.so" the ser couldn't start up and print the error message "ERROR while initialzing module auth_radius". I followed the procedure very carefully in http://www.iptel.org/ser/doc/ser_radius/ser_radius.html but this still happen. I don't know who to ask for. Is there any mailing list, its archive, or anyone that I can ask the question? Thanks, lin __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

Re: [Serusers] domain trouble
by George Lambson 30 Nov '05

30 Nov '05

Nevermind, I was putting the username only in the authorization user and it needs the username and the domain ie. bob(a)foobar.com. Thanks >>> Klaus Darilion <klaus.mailinglists(a)pernau.at> 11/29/2005 1:56 AM >>> The SIP_DOMAIN is only used for serctl as default domain (when you create a new user without specifiying the domain of the SIP AoR). You can specify the domain for which ser is responsibly (== myself) by using the alias=... parameter in ser.cfg, e.g.: alias=foo.com bar.com iptel.org mysipproxy.com Then you can verify using e.g. if (uri == myself) ... Another approach is using the domain module. Then the domains are stored in a DB table and you can add/delete domains without restarting ser. Then you have to use the functions provided my the domain module, e.g. if (is_uri_host_local()) ... regards klaus George Lambson wrote: > For some reason my SER isn't using the sip domain I specify (export SIP_DOMAIN="whatever.edu"). When my UA registers it will only register if I take out all of the digect authentication from my config file (I leave all of them blank instead of putting in a domain) and then use the host name of my server instead of a domain. > > Any ideas what I am doing wrong? > > Thanks, > George > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

Re: [Serusers] domain trouble
by George Lambson 30 Nov '05

30 Nov '05

Okay, that makes a lot of sense. But... I did it and changed the realm accordingly in my UA and now it won't authenticate with either realm. What now? >>> Klaus Darilion <klaus.mailinglists(a)pernau.at> 11/29/2005 1:56 AM >>> The SIP_DOMAIN is only used for serctl as default domain (when you create a new user without specifiying the domain of the SIP AoR). You can specify the domain for which ser is responsibly (== myself) by using the alias=... parameter in ser.cfg, e.g.: alias=foo.com bar.com iptel.org mysipproxy.com Then you can verify using e.g. if (uri == myself) ... Another approach is using the domain module. Then the domains are stored in a DB table and you can add/delete domains without restarting ser. Then you have to use the functions provided my the domain module, e.g. if (is_uri_host_local()) ... regards klaus George Lambson wrote: > For some reason my SER isn't using the sip domain I specify (export SIP_DOMAIN="whatever.edu"). When my UA registers it will only register if I take out all of the digect authentication from my config file (I leave all of them blank instead of putting in a domain) and then use the host name of my server instead of a domain. > > Any ideas what I am doing wrong? > > Thanks, > George > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Users] t_replicate after failure
by Douglas Garstang 30 Nov '05

30 Nov '05

In the scenario below, if the first t_replicate fails because 192.168.10.5 returns "Not Found", the second t_replicate command does not execute. Why? How can I make it execute? Call me crazy, but this doesn't seem at all intuitive to me. In a functional programming world, if the first fails, the next should still run. if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voip.com", "subscriber")) { www_challenge("voip.com", "0"); exit; }; save("location"); t_replicate("192.168.10.5","5060"); t_replicate("192.168.10.7","5060"); exit; }; Thanks, Doug.

1 0

[Serusers] Acc fields
by Richard C. Thompson 30 Nov '05

30 Nov '05

I hope someone can help answer this question. I am using Mysql acc table to generate cdr's and I am having problems with missing BYE records. I am using the totag field as a unique identifier to match up the INVITES, ACK and BYE's. Is this the field I should be using? I know ser changes the BYE uri to reflect the uac that issued the BYE but I don't know if it changes the totag. Thanks for your reply. Rick Thompson

2 1

[Users] Sending Authorisation Credentials
by Douglas Garstang 30 Nov '05

30 Nov '05

I have user agents registering with OpenSER. I wanted to 'copy' the registration to an Asterisk server, so I modified a standard openser.cfg as such: if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voip.com", "subscriber")) { www_challenge("voip.com", "0"); exit; }; save("location"); t_replicate("192.168.10.7","5060"); # Added this line... exit; }; I just stuck the t_replicate() call in there. OpenSER sends the REGISTER packet to the Asterisk box who requests credentials. How can I get OpenSER to send credentials (ie act as a UAC) for a given user, to a UAS? Thanks, Doug

2 1

[Serusers] CDRTool
by Martin Anderberg 30 Nov '05

30 Nov '05

(I know this is a bit out of topic, please dont feel offended) Hi there! I was about to try out CDRTool to see if it does what it says - however, it doesn't run at all. There is no forum (that i know of) that discusses CDRTool so i hope someone here can help. The box is a debian unstable (tried stable aswell) and CDRTools is installed according to the instructions. When i try to execute ie something in the scripts/-directory i get the following error: Warning: main(SOAP/Client.php): failed to open stream: No such file or directory in /var/www/CDRTool/rating_lib.phtml on line 5 Fatal error: main(): Failed opening required 'SOAP/Client.php' (include_path='.:/usr/share/php:/usr/share/pear') in /var/www/CDRTool/rating_lib.phtml on line 5 Ok, SOAP/Client.php doesn't exist, SOAP/client.php does however. A symlink ought to solve that problem? Running in the scripts/-directory again gives the error: #!/usr/bin/php4 Fatal error: Cannot redeclare class db_sql in /var/www/CDRTool/phplib/db_mysql.inc on line 13 Now i'm stuck and dont know what to do. If someone has any clues, please let me know. Thanks Martin

2 1

[Serusers] Issues with SER for TCP load testing
by Stephen McVarnock 30 Nov '05

30 Nov '05

Hi folks, I am new to the mailing lists - unsure as to which list is the most appropriate I have mailed to both - I apologise for this in advance! I am using SIPP and SER(v 0.9.4) to load test an application - SIPP is sending calls via UDP to SER at 15 cps and with a duration of 180 seconds. SER is then forwarding on the calls to the application using TCP and receiving the replies back over TCP. However I am seeing quite a few errors in SER after a while. I'd like to determine if the issues I am seeing are with SER or with the application itself so I'd appreciate some help with the following errors. Can someone tell me, for each of the error traces below, what is the root cause of it and (even better) how I can resolve it for SER - if it is indeed an issue with SER and not with the application. 1) 123(14715) ERROR: tcpconn_new: mem. allocation failure 123(14715) ERROR: tcp_main_loop: tcpconn_new failed, closing socket 2) 54(14646) ERROR: tsend_stream: failed to send: (32) Broken pipe 54(14646) ERROR: tcp_send: failed to send 3) 65(14785) ERROR: tcp_blocking_connect: SO_ERROR (146) Connection refused 65(14785) ERROR: tcpconn_connect: tcp_blocking_connect failed 65(14785) ERROR: tcp_send: connect failed 65(14785) msg_send: ERROR: tcp_send failed 4) 63(14783) ERROR: tsend_stream: send timeout (10000) 63(14783) ERROR: tcp_send: failed to send 5) There was also another error on the TCP select loop - select failed: Too many connections (or something similar). I have been playing around with the number of descriptors set using limit but without success. Like I said, my main concern is to eliminate SER where possible as the cause of these issues before focusing attention on the application. Thanks in advance, Steve.

2 1

[Serusers] Creating CDRs from SERs accouning records
by Adrian Georgescu 30 Nov '05

30 Nov '05

Frank, you can download CDRTool from: http://cdrtool.ag-projects.com/ CDRTool it is not publicly supported, we provide support only for our customers but you may give it a try. Adrian >>>>>>>>>>> Hi all i'm looking for a free/opensource application that generates CDRs (not XDRs, must not be rated) from the accounting data/events generated by SER (i think this is usually called normalization). I think this would have to be an application that is able to match INVITEs and BYEs and form CDRs with call start timestamp, call duration and so on. I had a look at the list of accounting software on the iptel.org website, but most of there applications are highly integrated accounting/billing solutions, far more than what i need (and also most of them seem to be commercial products). In the mailinglist archive i only found some references to CDRTool which seems also to be commercial. Obv. there must be a free version of CDRTool but neither i'm sure whether it implements exactly what i need so i could "grab it" for my own purposes nor could i find a place to download the free version). Does anyone know if there is such a kind of application available? Or any other idea how to deal with my requirements? Thanks a lot for your help Frank

1 0

[Serusers] REGISTER message headers verification
by Aisling O'Driscoll 30 Nov '05

30 Nov '05

Hi, I was wondering if someone could please cast their eyes over the register sent from one of my phones behind a nat and the 200 OK sent back from SER. There has been an issue with no audio between a natted client and a public client. I looked at the invite and noticed the private address was being used in the Contact header field and the c field in the sdp so obviously the headers are not being rewritten properly....I then checked the register messages to see if the phone was registering correctly - The message sequence is below. I notice the private header is shown but that would probably be correct seeing fix_nated_register doesnt actually change the header, it simply adds fields, is that right? i.e. thats why it replaced fix_nated-contact to be more RFC compliant Anyway its the contact header fields sent back in the 200OK that I would like verified - Should the private address be listed as one of the options here? I have gone back to the onsip feature callfwd script using mediaproxy to eliminate any errors due to changes I may have made The ONLY difference between the scripts is that I have if(uri != myself) { route(4); route(1); break; }; instead of #if (!is_uri_host_local()) { # if (is_from_local() || allow_trusted()) { # route(4); # route(1); # } else { # sl_send_reply("403", "Forbidden"); # }; # break; # }; as otherwise the phones wouldn't register. Many thanks, Aisling. SER: 157.190.74.152 Phone Private Address: 172.16.3.13 NAT Box: 84.203.148.14 U 84.203.148.14:5060 -> 157.190.74.152:5060 REGISTER sip:157.190.74.152:5060 SIP/2.0..Via: SIP/2.0/UDP 172.16.3.13;bran ch=z9hG4bK7543671c6ca1f07a..From: "Aisling 2092" <sip:2092@157.190.74.152:5 060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Contact: <si p:2092@172.16.3.13>..Call-ID: 78f22336c383663f@172.16.3.13..CSeq: 100 REGIS TER..Expires: 3456000..User-Agent: Grandstream BT100 1.0.6.7..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK7543671c6ca1 f07a;received=84.203.148.14..From: "Aisling 2092" <sip:2092@157.190.74.152: 5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Call-ID: 78 f22336c383663f@172.16.3.13..CSeq: 100 REGISTER..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.152:5060 " Noisy feedback tells: pid=1742 req_src_ip=84.203.148.14 req_src_port=5060 in_uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_cnt==1". ... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK754367 1c6ca1f07a;rport=5060;received=84.203.148.14..From: "Aisling 2092" <sip:209 2@157.190.74.152:5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5 060>;tag=4b358c93636b59b9f78eca99791ec991.cdd2..Call-ID: 78f22336c383663f@1 72.16.3.13..CSeq: 100 REGISTER..WWW-Authenticate: Digest realm="157.190.74. 152", nonce="438dfc0821fd70b65dbd9c0db3e5aa29fa1fcd82"..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.15 2:5060 "Noisy feedback tells: pid=1742 req_src_ip=84.203.148.14 req_src_po rt=5060 in_uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_ cnt==1".... # U 84.203.148.14:5060 -> 157.190.74.152:5060 REGISTER sip:157.190.74.152:5060 SIP/2.0..Via: SIP/2.0/UDP 172.16.3.13;bran ch=z9hG4bK9e94983538147a54..From: "Aisling 2092" <sip:2092@157.190.74.152:5 060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Contact: <si p:2092@172.16.3.13>..Authorization: Digest username="2092", realm="157.190. 74.152", algorithm=MD5, uri="sip:157.190.74.152:5060", nonce="438dfc0821fd7 0b65dbd9c0db3e5aa29fa1fcd82", response="90d926c009567996b99e76fef560b9c6".. Call-ID: 78f22336c383663f@172.16.3.13..CSeq: 101 REGISTER..Expires: 3456000 ..User-Agent: Grandstream BT100 1.0.6.7..Max-Forwards: 70..Allow: INVITE,AC K,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK9e9498353814 7a54;received=84.203.148.14..From: "Aisling 2092" <sip:2092@157.190.74.152: 5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>..Call-ID: 78 f22336c383663f@172.16.3.13..CSeq: 101 REGISTER..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.152:5060 " Noisy feedback tells: pid=1744 req_src_ip=84.203.148.14 req_src_port=5060 in_uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_cnt==1". ... # U 157.190.74.152:5060 -> 84.203.148.14:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.16.3.13;branch=z9hG4bK9e94983538147a54 ;rport=5060;received=84.203.148.14..From: "Aisling 2092" <sip:2092@157.190. 74.152:5060>;tag=2ec993227a53039b..To: <sip:2092@157.190.74.152:5060>;tag=4 b358c93636b59b9f78eca99791ec991.3e0a..Call-ID: 78f22336c383663f(a)172.16.3.13 ..CSeq: 101 REGISTER..Contact: <sip:2092@157.190.74.151>;expires=3279337, < sip:2092@172.16.3.13>;expires=3447275;received="sip:84.203.148.14:50 60", <s ip:2092@84.203.148.14:5060>;expires=3456000..Server: Sip EXpress router (0. 9.4 (i386/linux))..Content-Length: 0..Warning: 392 157.190.74.152:5060 "Noi sy feedback tells: pid=1744 req_src_ip=84.203.148.14 req_src_port=5060 in_ uri=sip:157.190.74.152:5060 out_uri=sip:157.190.74.152:5060 via_cnt==1".... ##exit 91 received, 0 dropped localhost:~ # -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Users] AVPOP and openser DB key
by Matteo Piazza 30 Nov '05

30 Nov '05

I use avpops module. I use the user_preferences table (openser DB) as avp_table: modparam("avpops","avp_table","usr_preferences") But I have a question: In the description of avp_table is not required a key, in the default configuration of user_preferences table there is a key composed by user-domain-attribute. But with this key I can't use some different value for the same attribute: for example "fwdbusy". So I have delete this key, and the program is ok. Someone knows why in the user_preferences the key is composed those three values -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 1

Re: [Users] How to Configure a Redirect Server Using OpenSer
by Bogdan-Andrei Iancu 30 Nov '05

30 Nov '05

Hi, do you get any error? do you see the 3xx reply? are you sure the script execution gets there? regards, bogdan Ajay Srivastava wrote: >Hi bogdan, >Thanks for ur suggestion. I made changes in openser.cfg as per the example >which you have given and then restart openser, but still not getting >redirect messages, should I need to do something more for enabling this >option? Please help me and tell me.. >Regards, >ajay > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Monday, November 28, 2005 5:02 PM >To: Ajay Srivastava >Cc: users(a)openser.org; Aviraj Saha >Subject: Re: [Users] How to Configure a Redirect Server Using OpenSer > >Hi Ajay, > >there is a script example on the cvs: > >http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/examples/redirect.c >fg?rev=1.2&view=auto > >regards, >bogdan > >Ajay Srivastava wrote: > > > >>Hi >> >>We are trying to configure OpenSer as Redirect Server. Can anyone >>provide me with a configuration file that explains what options we >>have in OpenSer to configure it as redirect server. >> >> >> >>With Regards >> >>Ajay >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >>

1 0

[Users] uac_replace_from errors
by Patrick Jordan-Smith 30 Nov '05

30 Nov '05

Hi there, I'm not totally sure if this is a problem with SER or not, but I have a interesting problem when using uac_replace_from together with LinkSys PAP2 ATAs (and possibly other linksys equipment, retesting tonight.) When you configure a name in the linksys ATA, you can put your name, ie, for myself I put Patrick Jordan-Smith. The linksys sends this like: From: Patrick Jordan-Smith <sip:728728@sip.qsi.net.nz> When I try to use uac_replace_from with this from address it chokes on it: ERROR: parse_to : unexpected char ["] in status 8: <<Beetlestone >> . ERROR:parse_from_header: bad from header ERROR:uac:restore_from_reply: failed to find/parse FROM hdr ERROR: parse_to : unexpected char ["] in status 8: <<Beetlestone >> . ERROR:parse_from_header: bad from header And I end up with a wrecked From address then nothing works. I have checked RFC 3261 and it only says quote marks should be used if the field contains a comma, question-mark or semi-colon. So it seems Linksys are following spec and SER is in error. Any comments on this? Thanks, Pat -- Patrick Jordan-Smith Network Operations Mercury Telecommunications Ltd / Quicksilver Internet voice +64-9-916-0300 / +64-27-5900595 fax +64-9-916-0301

4 5

[Users] PIC/SER PA presence module port?
by Joachim Fabini 30 Nov '05

30 Nov '05

Hi, Did anyone succeed (or, at least, try) to port the new SER pa module (Jamey Hicks' additions for PIC/SER trial) to openSER? If yes: any hints on how to do this? If not a question to the developers: Are the module interfaces of SER and OpenSER still so close that a port of the PA module should be feasible without too much coding? There is much interest in presence around and having a decent presence module in OpenSER is a definite plus. tia, --Joachim

3 3

[Users] How can I send radius authentication packet with openser
by Arda Tekin 30 Nov '05

30 Nov '05

Hi, I have installed openser, mysql, radiusclient-ng-0.5.2 successfully on REL3.0. openser works well with mysql. I need to send a radius authentication packet to a radius server(according to RFC2865). Packet contains base params: User-name (attr.1) $Username Password (attr.2) $Password NAS-Identifier (attr.4) (auto-generated) NAS-Port (attr.5) $uref State (attr.24) 0 Client-Port-DNIS (attr.30) NONE Caller-Id (attr.31) $calling I can not find a clear sample about radius. Which module is used for this purpose? Regards Arda

2 5

[Serusers] serctl via unix socket?
by Cesc 30 Nov '05

30 Nov '05

Hi, We use SER+SEMS in my project. Recently, we moved the interface between them to unix sockets (from the "traditional" fifo). But now ... serctl won't work because it cannot find the ser fifo (for example, "serctl ul show"). Can the two (fifo and unix sock) cohexist? that is ... can ser open both at the same time and receive/send from/to both? Or, how can i make serctl use the socket interface? Regards, Cesc

2 2

[Users] not going to failure_route
by Jayesh Nambiar 30 Nov '05

30 Nov '05

hi all, I was trying the following scenario: If a call comes from trusted source_ip, I rewritehost and forward them. But what I want is if the reply status is greater than 400, it should go to the failure route and try that destination. But the call gets replied as "500 service unavailable". I think, the call should go to the failure route if the status is greater than 300, is it right? Or i m making some logical error here. snippet of my cfg file: if (uri=~"^sip:00[0-9]*@") { #forward the trusted IPs without authentication strip(2); rewritehost("216.XX.XX.XX"); t_on_failure("1"); t_on_reply("2"); t_relay(); }; failure_route[1] { log(1, "trying next ip address"); prefix("00"); rewritehost("63.XX.XX.XX"); t_relay(); } onreply_route[2] { if(status=~"18[0-9]") { t_on_failure("0"); }; } pls help me out with this. thanx a lot in advance!! Jayesh --------------------------------- Enjoy this Diwali with Y! India Click here

2 1

Fwd: [Users] sample configuration file for redirect server
by jyotisankar nayak 30 Nov '05

30 Nov '05

HI Iqbal, pls help me in this matter. regards jyoti Note: forwarded message attached. --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

2 1

RE: [Users] Forwarding from dispatcher
by Douglas Garstang 30 Nov '05

30 Nov '05

Yes, I think the documentation needs an update. :) -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Tuesday, November 29, 2005 9:52 PM To: rick(a)eltopia.com Cc: users(a)openser.org Subject: Re: [Users] Forwarding from dispatcher On 11/29/05 23:15, Rick Richardson wrote: > I might be missing something obvious. But I have set up a dispatching > openSER box using the recommended .cfg file from the dispatcher module > docs and a custom list. > > On init, The parser complains of the forward(); function because it has > no params. I looked at the core docs and they say they forward() > requires a host and port, which makes sense, however, I don't know what > to put in as a param as it is decided by the ds_select_dst() function. > > In short, what do I pass to forward to make it use the destination > decided by ds_select_dst? > use "forward(uri:host, uri:port)" Cheers, Daniel > thanks, > > Rick > > > _______________________________________________ > 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

[Users] RE: Users Digest, Vol 6, Issue 98
by Papadopoulos Georgios 30 Nov '05

30 Nov '05

Lokesh, Comment lines start with # not ; George > Date: Wed, 30 Nov 2005 11:18:58 -0000 > From: "Lokesh Kumar" <lokesh(a)interacesso.pt> > Subject: Re: [Users] Can anyone share his openser.cfg..in which > openser is routing calls to pstn gateway > To: "Norman Brandinger" <norm(a)goes.com> > Cc: serusers(a)iptel.org, users(a)openser.org > Message-ID: <004901c5f59f$dbb87030$2f00a8c0@lokesh> > Content-Type: text/plain; charset="iso-8859-1" > > Hello Norman, > When I type openser start it gives me " ERROR: Bad config > file (4 errors). > here is the messages i got from cat /var/log/messages > --------- > Nov 30 11:05:01 ser crond(pam_unix)[26819]: session closed > for user root Nov 30 11:07:16 ser openser: ERROR: > load_module: could not open module > <modules/sl/sl.so>: modules/sl/sl.so: cannot open shared > object file: No such file or directory Nov 30 11:07:16 ser > openser: parse error (10,13-14): failed to load module Nov 30 > 11:07:16 ser openser: ERROR: load_module: could not open module > <modules/tm/tm.so>: modules/tm/tm.so: cannot open shared > object file: No such file or directory Nov 30 11:07:16 ser > openser: parse error (11,13-14): failed to load module Nov 30 > 11:07:16 ser openser: parse error (12,1-2): syntax error Nov > 30 11:07:16 ser openser: parse error (12,1-2): > Nov 30 11:09:11 ser openser: ERROR: load_module: could not open module > <modules/sl/sl.so>: modules/sl/sl.so: cannot open shared > object file: No such file or directory Nov 30 11:09:11 ser > openser: parse error (10,13-14): failed to load module Nov 30 > 11:09:11 ser openser: ERROR: load_module: could not open module > <modules/tm/tm.so>: modules/tm/tm.so: cannot open shared > object file: No such file or directory Nov 30 11:09:11 ser > openser: parse error (11,13-14): failed to load module Nov 30 > 11:09:11 ser openser: parse error (12,1-2): syntax error Nov > 30 11:09:11 ser openser: parse error (12,1-2): > [root@ser openser]# > ------------------------------ > > Here is my openser.cfg , In which i had commented mysql, and > auth modules.I want the things works without any AAA, initially. > > > > # > # $Id: pstn.cfg,v 1.2 2005/07/25 16:27:33 miconda Exp $ # # > example: ser configured as PSTN gateway guard; PSTN gateway > is located # at 192.168.0.10 # > > # ------------------ module loading ---------------------------------- > > loadmodule "modules/sl/sl.so" > loadmodule "modules/tm/tm.so" > ;loadmodule "modules/acc/acc.so" > loadmodule "modules/rr/rr.so" > loadmodule "modules/maxfwd/maxfwd.so" > ;loadmodule "modules/mysql/mysql.so" > ;loadmodule "modules/auth/auth.so" > ;loadmodule "modules/auth_db/auth_db.so" > ;loadmodule "modules/group/group.so" > loadmodule "modules/uri/uri.so" > > # ----------------- setting module-specific parameters --------------- > > ;modparam("auth_db", "db_url","mysql://ser:heslo@localhost/ser") > ;modparam("auth_db", "calculate_ha1", yes) > ;modparam("auth_db", "password_column", "password") > > # -- acc params -- > ;modparam("acc", "log_level", 1) > # that is the flag for which we will account -- don't forget > to # set the same one :-) ;modparam("acc", "log_flag", 1 ) > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > > /* ********* ROUTINE CHECKS > ********************************** */ > > # filter too old messages > if (!mf_process_maxfwd_header("10")) { > log("LOG: Too many hops\n"); > sl_send_reply("483","Too Many Hops"); > return; > }; > if (len_gt( max_len )) { > sl_send_reply("513", "Wow -- Message too large"); > return; > }; > > /* ********* RR ********************************** */ > > /* grant Route routing if route headers present */ > if (loose_route()) { t_relay(); return; }; > > /* record-route INVITEs -- all subsequent requests > must visit us */ > if (method=="INVITE") { > record_route(); > > > }; > > # now check if it really is a PSTN destination which > should be handled > # by our gateway; if not, and the request is an > invitation, drop it -- > # we cannot terminate it in PSTN; relay non-INVITE > requests -- it may > # be for example BYEs sent by gateway to call originator > if (!uri=~"sip:\+?[0-9]+@.*") { > if (method=="INVITE") { > sl_send_reply("403", "Call cannot be > served here"); > } else { > forward(uri:host, uri:port); > }; > return; > }; > > # account completed transactions via syslog > setflag(1); > > # free call destinations ... no authentication needed > if ( is_user_in("Request-URI", "free-pstn") /* free > destinations */ > | uri=~"sip:[79][0-9][0-9][0-9]@.*" > /* local PBX */ > | uri=~"sip:98[0-9][0-9][0-9][0-9]") { > log("free call"); > } else if (src_ip==212.xx.35.xx) { > # our gateway doesn't support digest authentication; > # verify that a request is coming from it by source > # address > log("gateway-originated request"); > } else { > # in all other cases, we need to check the > request against > # access control lists; first of all, verify request > # originator's identity > > if (!proxy_authorize( "gateway" /* realm */, > "subscriber" /* table name */)) { > proxy_challenge( "gateway" /* realm > */, "0" /* no qop */ ); > return; > }; > > # authorize only for INVITEs -- RR/Contact > may result in weird > # things showing up in d-uri that would break > our logic; our > # major concern is INVITE which causes PSTN costs > > if (method=="INVITE") { > > # does the authenticated user have a > permission for local > # calls (destinations beginning with > a single zero)? > # (i.e., is he in the "local" group?) > if (uri=~"sip:0[1-9][0-9]+@.*") { > if > (!is_user_in("credentials", "local")) { > sl_send_reply("403", > "No permission for local calls"); > return; > }; > # the same for long-distance > (destinations begin with two zeros") > } else if (uri=~"sip:00[1-9][0-9]+@.*") { > if > (!is_user_in("credentials", "ld")) { > sl_send_reply("403", > " no permission for LD "); > return; > }; > # the same for > international calls (three zeros) > } else if (uri=~"sip:000[1-9][0-9]+@.*") { > if > (!is_user_in("credentials", "int")) { > sl_send_reply("403", > "International permissions needed"); > return; > }; > # everything else (e.g., > interplanetary calls) is denied > } else { > sl_send_reply("403", "Forbidden"); > return; > }; > > }; # INVITE to authorized PSTN > > }; # authorized PSTN > > # if you have passed through all the checks, let your > call go to GW! > > rewritehostport("212.xx.35.xx:5060"); > > # forward the request now > if (!t_relay()) { > sl_reply_error(); > return; > }; > > } > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

1 0

[Serusers] Error opening ser's FIFO /tmp/ser_fifo
by subbanarasa 30 Nov '05

30 Nov '05

/Hi, // //I had installed already the version 0.8.12 running at a MySql SERVER. The service ////at this moment, STARTS normally with no interruption but when I do use the command line : ////"serctl + [any option] - I got the error : "Error opening ser's FIFO /tmp/ser_fifo".Make sure you have line fifo=/tmp/ser_fifo in your config but ser.config file like this # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- 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" # ------------------ 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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_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) modparam("usrloc", "db_url", "mysql:/ser:reddy@10.232.19.169/ser") # -- 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{ # 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("10.232.19.169", "subscriber")) { www_challenge("10.232.19.169", "0"); break; }; save("location"); 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(); }; } /

1 0

Re: [Serusers] SER and Asterisk authentication
by Voipers Portugal 30 Nov '05

30 Nov '05

Will it work with SER or only for OpenSER? I'm sorry but I don't know the difference beteween those 2. Jose Simoes On 11/28/05, Jim Dalton <jim.dalton(a)transnexus.com> wrote: > > One alternative to consider is a certificate authority/peering server that > issues a digitally signed token authorizing each call between SER and > Asterisk. This eliminates the threat of IP spoofing and the need top > maintain IP access lists on each SER and Asterisk box. See > http://osp-module.berlios.de/. > > Jim Dalton > > > ------------------------------ > *From:* serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] *On > Behalf Of *Voipers Portugal > *Sent:* Friday, November 25, 2005 2:21 PM > *To:* sip(a)perceval.net > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] SER and Asterisk authentication > > > Thanks, i'll read it now. But this only solves one part of the problem, > that is, Asterisk can authenticate in SER. And how SER can authenticate in > Asterisk? Is it possible only by IP? > > Jose Simoes > > > On 11/25/05, Arek Bekiersz <sip(a)perceval.net> wrote: > > > > OK, I understand your problem. > > Please read <carefully> Asterisk Wiki for sip.conf at: > > > > http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf > > > > > > To register, you just need to insert this into general section of > > Asterisk's sip.conf: > > > > [general] > > ; Register with a SIP provider - register username at SER as > > local_asterisk_extension at Asterisk > > > > register => > > username:digest_password@your.ser.server/local_asterisk_extension > > > > [your.ser.server] > > type=friend > > host=your_ser.server > > username=username > > > > > > -- > > Arek > > > > > > > > Voipers Portugal wrote: > > > But what do I have to add? I can add a user, with username Asterisk, > > but > > > how can I associate that user to the INVITE message from Asterisk to > > the > > > SER? Don't understand how SER can authenticate if i cannot associate > > the > > > RADIUS/LDAP user to the Asterisk box. > > > > > > Jose Simoes > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > >

1 0

[Serusers] JAIN SIMPLE stack?
by Andrey Kouprianov 30 Nov '05

30 Nov '05

Hi! Does anyone know, when will JAIN IM and Presence (JSP164 and JSR165) implementaions become available? Is Panasonic the one to make it?

1 0

[Serusers] SER Bug? 503 messsage is not handled properly
by Pat wang 30 Nov '05

30 Nov '05

I have a test setup like: UAC1-----SER------Third Party Proxy----UAC2 UAC1 registers to SER directly while UAC2 registers to SER via a third party proxy. When UAC1 calls UAC2, the third party proxy replies a 100 trying then 503 Service Unavailable to SER. There is no Retry-After in the 503 message. SER keeps re-sending INVITE. By RFC, If no Retry-After is given, the client MUST act as if it had received a 500 (Server Internal Error) response. Is this a bug on SER? Thanks, Pat

1 0

RE: [Serusers] Local ACKs for SER-SEMS
by Bjorn Asmul 30 Nov '05

30 Nov '05

I'm using latest stable SER (0.9.5.pre2), and we have problems with a Grandstream GXP-2000 phone with firmware 1.0.1.9. We are trying to initiate a call from a Cisco gateway via SER to the GXP. SER tries to set up the call, but there seem to be an ACK missing somewhere, so the Cisco keeps retrying the call. >From the trace I could see this message: sl_filter_ACK: to late to be a local ACK! Would this patch fix this problem? Also I did a re-compile from CVS (rev. rel_0_9_0) today. Would this patch have been applied? Thanks, Bjorn -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of SER.Jan Janak Sent: Tuesday, November 29, 2005 7:42 AM To: Cesc Cc: OpenSER-dev; SER-Users Subject: Re: [Serusers] Local ACKs for SER-SEMS Great, thanks a lot. I just commited your fix. Jan. On 28-11-2005 18:51, Cesc wrote: > Hi, > > I found it ... now it works. There is a pseudo-patch at the end of the > message (pseudo because it cannot directly be applied ... it is just > to mark the three lines i needed to change ... ). The patch is for > ser-rel-0.9.4 (i send it to openser-devel ... i did not look, but > probably the problem is there too). > > It is just a small bug ... from 0.9.0 to 0.9.4 a part of the loop was > moved outside the loop, to the end of the function ... and in a place > where there used to be a return now the loop would just continue ... > Then, the matching_3261 function would, if a transaction matching the > ACK was found, would always return e2e ack ... even though the > ack_matching function returned it as a local transaction ... > > For me, this patch fixes it ... but as it is a delicate part, i am not > sure if it affects other parts ... i don't think so, but ... > > Tks for all who took a look at the problem (greger, jan, samuel, ... )! > > Regards, > > Cesc > PS - This should affect anyone using SEMS ... so, if no one saw it ... > does it mean there is really a few people using it? what a pitty ... > > > Index: modules/tm/t_lookup.c > =================================================================== > --- modules/tm/t_lookup.c (revision 8) > +++ modules/tm/t_lookup.c (working copy) > > @@ -292,9 > struct via_body *via1; > int is_ack; > int dlg_parsed; > - int ret; > + int ret = 0; > struct cell *e2e_ack_trans; > > e2e_ack_trans=0; > via1=p_msg->via1; > is_ack=p_msg->REQ_METHOD==METHOD_ACK; > > @@ -327,16 > if (is_ack && p_cell->uas.status<300 && e2e_ack_trans==0) { > /* make sure we have parsed all things we need for dialog > * matching */ > if (!dlg_parsed) { > dlg_parsed=1; > if (!parse_dlg(p_msg)) { > LOG(L_ERR, "ERROR: matching_3261: dlg parsing failed\n"); > return 0; > } > } > ret=ack_matching(p_cell /* t w/invite */, p_msg /* ack */); > if (ret>0) { > e2e_ack_trans=p_cell; > + break; > } > /* this ACK is neither local "negative" one, nor a proxied > * end-2-end one, nor an end-2-end one for a UAS transaction > > @@ -358,9 > > /* just check if it we found an e2e ACK previously */ > if (e2e_ack_trans) { > *trans=e2e_ack_trans; > - return 2; > + return ret; > } > DBG("DEBUG: RFC3261 transaction matching failed\n"); > return 0; > } _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] How to Implement 3 way call confrence in SER
by sagar 30 Nov '05

30 Nov '05

Hi All, Is it possible to Implement 3 way call confrence using SER. I am using ser 0.8.14 with asterisk b2bua. If it is possible from where can i get the help or config files. Thanks in advance Vidhya sagar Dixit --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

4 3

[Serusers] Repeated Register Message
by tcchan 30 Nov '05

30 Nov '05

Hi All, I encountered a situation that somehow a UA keep on Registering with the server with the same IP but on different port. So SER, thought that there are multiple instances of the same username and save the various location into the database. The problem is it registers so often that the memory is full not very long. I suspect it has something to do firewall but I am not sure. Has anyone encounter this problem before? Any hint?? For the time being, is there a way to limit the number of REGISTER message that the SER will take? Regards, TC Chan

3 2

[Users] sample configuration file for redirect server
by jyotisankar nayak 30 Nov '05

30 Nov '05

hi, i want to configure the openser to behave as a redurect server i.e when i send a request to the openser server it send back the request to client with zero or more addresses in the contact field. so can anybody pls send a sample configuration file which will force openser to behave as a redirect server. regards, jyoti sankar nayak. --------------------------------- Yahoo! DSL Something to write home about. Just $16.99/mo. or less

1 0

[Users] Forwarding from dispatcher
by Rick Richardson 30 Nov '05

30 Nov '05

I might be missing something obvious. But I have set up a dispatching openSER box using the recommended .cfg file from the dispatcher module docs and a custom list. On init, The parser complains of the forward(); function because it has no params. I looked at the core docs and they say they forward() requires a host and port, which makes sense, however, I don't know what to put in as a param as it is decided by the ds_select_dst() function. In short, what do I pass to forward to make it use the destination decided by ds_select_dst? thanks, Rick

2 1

[Users] error when i make a call!
by messa＠innsof.com 30 Nov '05

30 Nov '05

Hi all I have these errors when running my openser system: 0(21323) ERROR: t_newtran: transaction already in process 0xb616ba38 0(21323) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) What do you think about it? Best regards! Serge

2 1

[Serusers] VoIP provider for SER?
by Dejan Cc 30 Nov '05

30 Nov '05

Dear friends, My company have used SER as backbone of our IP telephony system and Asterisks at subscriber's sites . Asterisks register to SER. Now, we would like to use SER to originate our international traffic to VoIP provider that would support such solution (connections from SER). Is there any VoIP provider that could provide us such service? If anyone knows any, please, help us. Any reply on this issue will be greatly appreciated. Best regards Dejan Cc

2 1

[Serusers] SER INVITE 200OK retransmission timer
by hpvpl 29 Nov '05

29 Nov '05

Hi, I'm performing performance test using SER 0.8.12 on linux red-hat with a default ser.cfg file (the one provided with installed package). I can't find in the documentation the default values used for SIP INVITE and 200OK retransmission. What are these default value ? Thanks for the support Cheers

1 0

[Serusers] Modules mysql
by Marcel van Kaam, Fonetica Belgie 29 Nov '05

29 Nov '05

Hi All, Just checked on the list, found already some answers but still no luck. I use the getting started files from Onsip.org. Running Debian 2.6.12 ERROR: module modules/cpl/cpl.so not compiled ERROR: module modules/cpl-c/cpl-c.so not compiled ERROR: module modules/extcmd/extcmd.so not compiled ERROR: module modules/jabber/jabber.so not compiled ERROR: module modules/mysql/mysql.so not compiled When I use make all; make install, this is the outcome of my SER installation. I also tried make module etc. Is Debian the problem or do I need to make some changes to the Makefiles? Any help is welcome. Marcel

1 1

Re: [Serusers] Cisco 7960 SIP firmware
by Ladislav Andel 29 Nov '05

29 Nov '05

Thank you for pointing me at. I will look at it. Ladislav Atle Samuelsen wrote: > offtopic, and never said by me... > > broadvoice in USA has a tftpserver.. with the images on.. just point > your phone to it.. and wWops it's upgraded to atest SIP-image :-) > (ps, I've never done it myself) > > -A > > * Ladislav Andel <ladia6(a)centrum.cz> [051129 20:27]: > >> Hello, >> I'm sorry for this off topic but would anybody know where can I get SIP firmware for Cisco 7960 IPphone? >> What is the price or can anybody share, please? I'm speaking for my professor and am trying to get the firmware >> for our lab IP phones for the lab purposes, not any commercial use. It would help even me to complete and show off my results >> of my semester project. >> >> Thank you for any hints or redirecting. >> >> Ladislav >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > > >

1 1

[Serusers] Cisco 7960 SIP firmware
by Ladislav Andel 29 Nov '05

29 Nov '05

Hello, I'm sorry for this off topic but would anybody know where can I get SIP firmware for Cisco 7960 IPphone? What is the price or can anybody share, please? I'm speaking for my professor and am trying to get the firmware for our lab IP phones for the lab purposes, not any commercial use. It would help even me to complete and show off my results of my semester project. Thank you for any hints or redirecting. Ladislav

1 0

[Users] Is anyone using xtunnels successfully with openser....???
by Lokesh Kumar 29 Nov '05

29 Nov '05

Hello List, I just want to know if any one has been configured xtunnels successfully with openser and freeradius. Regards

1 0

[Serusers] How to Implement 3 way call confrence in SER.
by sagar 29 Nov '05

29 Nov '05

Hi All, Is it possible to Implement 3 way call confrence using SER. I am using ser 0.8.14 with asterisk b2bua. If it is possible from where can i get the help or config files. Thanks in advance Vidhya sagar Dixit --------------------------------- Yahoo! DSL Something to write home about. Just $16.99/mo. or less

1 0

[Users] postgres and openser 1.0.0
by JF 29 Nov '05

29 Nov '05

Hi, I'm trying to use postgres module with auth_db module in openser 1.0.0 but it is faling to load usrloc module. I get the following error: 0(0) find_mod_export: <db_use_table> in module mysql not found 0(0) bind_dbmod: Module mysql does not export db_use_table function 0(0) ERROR: mod_init(): Can't bind database module 0(0) init_mod(): Error while initializing module usrloc but I'm not using mysql module! In config file I have: # ------------------ module loading ---------------------------------- #loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/mysql.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/postgres.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/sl.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/tm.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/rr.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/usrloc.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/registrar.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/auth.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/auth_db.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/xlog.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/uri.so" loadmodule "/usr/local/openser-1.0.0/lib/openser/modules/lcr.so" Why is it searching in mysql module if I'm using postgres? Any help appreciated. JF

2 2

Re: [Serusers] Re: [Users] rfc3264 and nathelper module
by sip 29 Nov '05

29 Nov '05

On Tue, 29 Nov 2005 14:46:55 +0100 (CET), harry gaillac wrote > > REGISTER: > > if (!is_from_local()) { > sl_send_reply("401", "Unauthorized"); > break; > }; > rewritehostport("nxs.yi.org:5050"); > t_relay_to_udp("nxs.yi.org","5050"); So... you want your users to register both with Asterisk AND with SER? What for? Why not use some form of SER authorisation to control access to the Asterisk server? I'm not sure what you're trying to do here... > > INVITE: > rewritehostport("nxs.yi.org:5050"); What's the context around this? You want ALL calls to go straight to Asterisk? It just seems... I don't know... weird. I assume you're doing this because SER handles presence and Asterisk doesn't... but everything else you want to be controlled by Asterisk, including user registration, etc, etc. For your invite, btw, you would need to also have a t_relay_to_udp, or alternatively, you can do it as: rewritehostport("nxs.yi.org:5050"); forward(uri:host,uri:port); break; > > sip.conf: > [general] > context=local ; Default context for > incoming calls > ; if asterisk was > compiled with OSP support. > realm=nxs.yi.org ; Realm for digest > authentication > ; defaults to > "asterisk" > ; Realms MUST be > globally unique according to RFC 3261 > ; Set this to your > host name or domain name > bindport=5050 ; UDP Port to bind to > (SIP standard port is 5060) > bindaddr=nxs.yi.org ; IP address to bind > to (0.0.0.0 binds to all) > srvlookup=yes ; Enable DNS SRV > lookups on outbound calls > tos=lowdelay ; > lowdelay,throughput,reliability,mincost,none > maxexpirey=3600 ; Max length of > incoming registration we allow > defaultexpirey=1000 ; Default length of > incoming/outoing registration > allow=all ; First disallow all > codecs > musicclass=default ; Sets the default > music on hold class for all SIP calls > language=fr ; Default language > setting for all users/peers > rtptimeout=60 ; Terminate call if 60 > seconds of no RTP activity > tpholdtimeout=300 ; Terminate call if > 300 seconds of no RTP activity > useragent=Asterisk PBX ; Allows you to change > the user agent string > dtmfmode = rfc2833 ; Set default dtmfmode > for sending DTMF. Default: rfc2833 > promiscredir = no ; If yes, allows 302 > or REDIR to non-local SIP address > > [84] > type=friend > secret=84 > username=84 > context=local > host=dynamic > mailbox=84 > allow=all > > [85] > type=friend > secret=85 > username=85 > context=local > host=dynamic > mailbox=85 > allow=all > In your blocks there, you might want to add insecure=very to allow registered hosts to call without reauthenticating -- just to speed things up a little (optional) one IMPORTANT thing you want to add to each block just to make sure it's there is: canreinvite=yes This allows the RTP traffic to bypass asterisk and just go from UA to UA without taking up bandwidth. It will NOT, however, work well for NATted users without playing around with rewriting the SDP (using something like a nathelper fix_nated_sdp("3") call) Rereading your original emails, I now think I understand that this is what you were asking about before (correct me if I'm wrong)... was the fix_nated_sdp not working correctly in some way? N.

1 0

[Users] cpl scripts and pstn gateway authentication
by Günther Starnberger 29 Nov '05

29 Nov '05

hello, what is the usual method to use the cpl module when a pstn gateway needs to know which user is responsible a call? e.g. i have a pstn gateway pstn.example. numbers are dialed by specifying them as user part of a sip uri => 00321234(a)pstn.example. as the gateway is an asterisk box i'm rather flexible in choosing a method which specifies how the name of the remote-user is transfered to the gateway. if a user does a forward to the pstn (e.g. in the case of no-answer) he should also be accounted for the call - so i either need to set the 'From' header to the owner of the cpl script or i need to add another header like 'X-account' which specifies the owner. a solution for this seems to be to store the name of the callee in an AVP before executing the script and to set the From field in proxy_route after the script has started. are there better or more-standard solutions for this problem? bye, /gst

2 4

[Users] LCR gw matching
by JF 29 Nov '05

29 Nov '05

Hi, Can anyone help clarify the following: According to LCR module docs for version 1.0.0, "gateway selection is based on caller's RPID URI (if available in caller's RPID AVP after authentication) or From URI and user part of Request-URI (telephone number)." What is the RPID AVP and how is it set? The docs also say "a gateway matches a request if user part of Request-URI and "From" URI match the prefix and From pattern of the gateway". But in gw table, there is no From pattern column, just prefix: gw_name | grp_id | ip_addr | port | uri_scheme | transport | prefix Also, it would be nice if gw matching could be based also on request method, request headers, or even arbitrary AVPs which would be set somewhere else. What do folks think about this? Thanks, JF

2 1

[Users] rfc3264 and nathelper module
by harry gaillac 29 Nov '05

29 Nov '05

Hello, Does nathelper module would support rfc3264 to provide MOH ? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

10 19

[Serusers] domain and "myself" question
by Kyriakos Mavromichalis 29 Nov '05

29 Nov '05

Hi, currently i have a configuration that works when UA use ip adresseses for contacting the server. Also ip is used for the domain. What i would like users to do is use a domain ie sip1.otenet-telecom.net to register. Domain is setup A record and points to the ip of the server. Users register ok but when they try to make a call, if(uri!=myself) returns true and the call does not go through. uri is number(a)sip1.otenet-telecom.net sip1.otenet-telecom.net is in the domain mysql table. What do i do to get the server recognize the calls from users using the domain? thanks -- Kyriakos Mavromichalis Otenet Telecom

6 7

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users