- sr-users - kamailio.org

Re: [Users] question false or true
by Daniel-Constantin Mierla 15 Mar '07

15 Mar '07

Hello, not sure I understood your question, but if you are looking to a way to mark an account enabled/disabled you can use groups or user preferences to achieve that. Cheers, Daniel On 03/15/07 22:22, Jobson Andrade wrote: > > Hello, > > > > I have one costumer and I need one feature in openser, more I no have > the response, hi need one form to able and disable one account (false > or true) in openser, the question is: > > > > This is possible, yes or no? > > > > Thanks in advanced > > > > > > Jobson Andrade > > > > Projetos & Desenvolvimento > Obelisk - The Asterisk & VoIP Experts > > > > phone/fax: (11) 2164-4808 ext. 115 > cell Phone: (11) 8175-9916 / 8271-0480 > email: jandrade(a)obelisknet.com.br <mailto:jandrade@obelisknet.com.br> > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Users] help in uac config
by Jobson Andrade 15 Mar '07

15 Mar '07

Hello, I need one help in my configuration using authentication Bellow is my configuration more not work, i need one help for work this configuration modparam("uac","credential","username:itsp.com.br:password") ##Send Call For DDI if (uri=~"^sip:00[1-9][1-9][0-9]*@") { route(3); exit; } route[3]{ #enviando route to hitconferencing strip(2); uac_replace_from("credential"); route(1); exit; } This is correct? Thanks in advancend Jobson Andrade Projetos & Desenvolvimento Obelisk - The Asterisk & VoIP Experts phone/fax: (11) 2164-4808 ext. 115 cell Phone: (11) 8175-9916 / 8271-0480 email: <mailto:jandrade@obelisknet.com.br> jandrade(a)obelisknet.com.br

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[Users] question false or true
by Jobson Andrade 15 Mar '07

15 Mar '07

Hello, I have one costumer and I need one feature in openser, more I no have the response, hi need one form to able and disable one account (false or true) in openser, the question is: This is possible, yes or no? Thanks in advanced Jobson Andrade Projetos & Desenvolvimento Obelisk - The Asterisk & VoIP Experts phone/fax: (11) 2164-4808 ext. 115 cell Phone: (11) 8175-9916 / 8271-0480 email: <mailto:jandrade@obelisknet.com.br> jandrade(a)obelisknet.com.br

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[Users] Presence information expires at presence server
by Rajaram Tripathy 15 Mar '07

15 Mar '07

Hi, Below is my scenario, User1 is the watcher and User2 is the presentity User1 sends SUBSCRIBE message to watch the presence inforamation of User2. User2 sends PUBLISH message, User1 gets notification. After first PUBLISH User2 does not refreshes its publication. Currently On expiry of presence information OpenSer sends NOTIFY with status as "closed" in the first tuple. Is this behaviour is correct ? Do we need to send NOTIFY with status as "closed" for each "tuple" ? What is the correct behaviour ? Also when we send PUBLISH with Expire header as "0", OpenSer sends NOTIFY with the XML body as it is in the first PUBLISH. Is it right behaviour ? Thanks in advance. Regards, Rajaram The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[Serusers] RTPproxy problem
by Joao Pereira 15 Mar '07

15 Mar '07

Hello I whant to run RTPproxy separatly from my SIP router, so I tried to run it like this: [root@vox rtpproxy-0.3]# rtpproxy -s udp:xxx.xxx.xxx.xxx:22222 rtpproxy: explicit binding address has to be specified in UDP command mode [root@vox rtpproxy-0.3]# rtpproxy -l xxx.xxx.xxx.xxx: -s udp:xxx.xxx.xxx.xxx::22222 xxx.xxx.xxx.xxx: = my IP But when I run SER, this happens 9(15400) WARNING: rtpp_test: support for RTP proxy has been disabled temporarily 11(15407) ERROR: send_rtpp_command: can't connect to RTP proxy I already tried with modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") and with modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock udp:xxx.xxx.xxx.xxx:22222") But it always says: send_rtpp_command: can't connect to RTP proxy Whats the problem? Im using this configuration in sip.conf: http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html Regards Joao Pereira

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[Users] in openser's website...
by Bodin Bruno 15 Mar '07

15 Mar '07

/*About */ FLASHNEWS : /*in first news : 2007-02-14*/ => /*2007-03-14, No ? Nevertheless, Really thank for OpenSer , It really a good soft :) Bodman */

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[Serusers] Pear and serweb
by POON Ben 15 Mar '07

15 Mar '07

Hi all, I'm trying to get serweb running. I am using Sparc Solaris 10, php 4.3.2, ser 0.9.3, serweb 0.9.4. I get this error when I try to go to the serweb page: Warning: main(DB.php): failed to open stream: No such file or directory in /usr/local/share/serweb/html/main_prepend.php on line 38 Fatal error: main(): Failed opening required 'DB.php' (include_path='.:/opt/sfw/lib/php') in /usr/local/share/serweb/html/main_prepend.php on line 38 Does serweb require Pear even if I am not using ldap? Or do I have to install it anyway? Or is that even Pear related? Can someone please straighten me out? I'm a little confused. Thank you very much :) Ben

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Re: [Users] openser+postgresql
by 赵俊杰 15 Mar '07

15 Mar '07

Hello,klaus First thank for your suggest! My OS I FC4,postgresql client is 8.0.3,openser 1.2.0 I try to update libpq to libpq4.It cann't work when I start the opener.The postgres.so is cann't be loaded. But using the mysql it works well. Before now when I execute openser_postgresql.sh it occurs the following error ~./pgpass does not exist, please create this file and support proper credentials for user postgres. Note: you need at least postgresql>= 7.3 And I write a .pgpass file to solve the problem.Following is the content of file: *:*:*:postgres:"" Do you have other suggestion? Thank you! Best Regards Junjie Zhao 2007-03-15

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RE: Re: [Users] SIP supplementary service
by 김형선 15 Mar '07

15 Mar '07

thanks. I took a look at onsip.org. But I could not find script files for other supplementary service. There are config files for 'getting started' and just 'call forward'. Are there config files for other supplementary services? Can OpenSER support supplementary services? thank you. -----Original Message----- From: "Bogdan-Andrei Iancu"<bogdan(a)voice-system.ro> To: "김형선"<liotliol(a)naver.com> Cc: users(a)openser.org Sent: 07-03-15(목) 02:44:58 Subject: Re: [Users] SIP supplementary service Hi, you may take a look at onsip.org for more samples. regards, bogdan 김형선 wrote: > > Hi. > > > > I saw the example script of openser.cfg for 'call forwarding'. > > Are other SIP supplementary services possible with OpenSER, by script? > > (call parking, call pickup, hold, conference, etc...) > > > > If it is possible, how can I get these script files? > > > > thanks > 새로운 기부문화의 씨앗, 해피빈

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[Users] auth_db and load_credentials question
by Klaus Darilion 15 Mar '07

15 Mar '07

Hi! When I load AVPs during authentication - what type is the content of the AVP? string? integer? Is it automatically derived from the column type (like avp_db_query)? My problem is: I want to load an integer value and compare it against an integer: modparam("auth_db", "load_credentials", "s:maxcalls=maxcalls") ... if(avp_check("$avp(i:40)", "ge/$avp(s:maxcalls)/g") { Does this work? What happens if I compare an integer with a string - will there be a conversion? thanks klaus -- Klaus Darilion nic.at

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[Users] Question about changing an coming call destination.
by Howard Tang 15 Mar '07

15 Mar '07

Hi All, I am having a problem to do the following. A call recevied from PSTN Caller is 1234 Callee is 5678 Normally, the UA 5678 that registered on the openser will receive the call and have caller id display 1234. What i want to do is the UA 1234 will recevice the call instead 5678, i don't care about the callid (basically, caller id will remain 1234). where should i preform the modification? I dont need Callerid. so i just want to send the incoming call to the account that match the caller id. My logic will be if sip_to = 5678, then change sip_to(5678) to sip_from(1234). send the call to the user 1234. However, i am not familiar with the code. Can someone show me how i can do that please? Regards, Howard

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[Serusers] Max number of routing blocks in ser 0.9.6
by Cesc 15 Mar '07

15 Mar '07

Hi, I just hit the max number of routing blocks in my config file ... I use the route blocks as "functions", to have a cleaner config or so I hope. The problem is that in config.h, RT_NO is set to 20 ... so, max num of routing blocks is 20. Can I just change it? or will it have unexpected side effects? Tks! Cesc

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Re: [Users] openser +ipv6
by 赵俊杰 15 Mar '07

15 Mar '07

Hello,Bernard Thank you very much! I follow you suggestion.Add listen=udp:[ipv6_adresss]5060 in the file opener.cfg.when I restart the service it works. Best regards Junjie Zhao 2007-03-15

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[Users] OpenSER 1.2 debs available for Debian stable
by Bogdan-Andrei Iancu 15 Mar '07

15 Mar '07

Hi, packages for Debian stable were uploaded both on the web download section http://www.openser.org/pub/openser/latest/packages/deb-stable/ and on the project debian repository deb http://www.openser.org/debian sarge main Regards, bogdan

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[Users] Re: Info Permissions - Trusted IP
by Dan-Cristian Bogos 15 Mar '07

15 Mar '07

Hi, my fault there, I was checking as ussually on the wrong place. Trusted module based on ip works fine no matter of the port of the originator. Cheers, Dan

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RE: [Users] Openser and Oracle
by Papadopoulos Georgios 15 Mar '07

15 Mar '07

I see what you mean. However, how is this avoided if you use query() instead of raw_query()? Function query() will also write the same string as the value of the User-Agent column. Both functions will construct the same query and at the end they will call submit_query(). No? best regards George > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, March 14, 2007 2:28 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Openser and Oracle > > Papadopoulos Georgios wrote: > > What do you mean by "SQL injection"? I know it is not a pretty > > solution (especially changing the code) but I cannot think > of an alternative. > > > Just an example. The save() function writes the User-Agent > header into the DB. If the usrloc would use a raw SQL query, > and the User-Agent header would be something like > User-Agent: foobar'; delete from subscriber; ' > > then the delete query would be executed. This is an SQL injection. > > regards > klaus > > > > > I can look into SER's oracle module. But that would mean > that in the > > where clauses all string values should be surounded with > "upper". Not > > a pretty solution either but it would work I guess. > > > > I briefly saw that Openser 1.2 has string transformations. So could > > uppercase and lowercase be included there? I am picturing something > > like avp_db_load("$ruri{s.lowercase}", ...) Of course this > would only > > solve the issue with avpops. In order to solve the issue > with auth_db > > for example, wouldn't we need new functions? > > > > Any comments about the multiple connections to the DB? > > > > > >> -----Original Message----- > >> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > >> Sent: Wednesday, March 14, 2007 12:59 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] Openser and Oracle > >> > >> Papadopoulos Georgios wrote: > >>> Hello, > >>> > >>> First of all congratulations to everyone involved in the > >> new release. > >>> I haven't been able to get my hands on it yet, but just > >> reading about > >>> it makes me very excited. Great job! > >>> > >>> I would like to address the issue of Openser and Oracle working > >>> together. We are currently using Openser with a local > >> MySQL. Our main > >>> database is Oracle and we are just copying data to MySQL so that > >>> Openser can work. This is a little difficult to maintain so > >> I thought > >>> I would try to make Openser use directly our Oracle. Of > >> course I ran > >>> into a number of issues. > >>> 1. modules/acc and unixodbc. The acc_db_request() was not working > >>> because inserting a string in a date column does not work > >> with Oracle. > >>> So, I had to change functions time2str() and time2odbc() in > >> order to > >>> make this work. Also had to change acc.c to treat column time as > >>> DB_TIME instead of DB_STR (I think this could be treated > as a bug). > >>> 2. modules/lcr. The query uses char_length() and rand() > >> which I had to > >>> replace with lengthc() and dbms_random.value. Should these > >> be modules > >>> params? > >> Hi! > >> > >> I guess oracle allows the defintion of new function. Then > you could > >> write the functions char_length() and rand() which would be just > >> wrappers to lengthc() and dbms_random.value. > >> > >>> 3. modules/avpops. All issues were resolved by config changes and > >>> replacing avp_db_load() with avp_db_query(). > >> Be careful - raw queries are vulnerable to SQL injection!!! > >> > >>> 4. modules/auth_db, alias_db, uri_db, group. Since MySQL is case > >>> insensitive and Oracle is not, I made changes in the code to use > >>> raw_query() instead of query(). All queries have to be in > the form > >>> "select ... from ... where username=upper(...)" > >> Again: Be careful - raw queries are vulnerable to SQL injection!!! > >>> > >>> I would be happy to provide patches and help in any way > in order to > >>> make this migration easier in the future. However I am not sure > >>> whether my changes are general enough for everybody to use. > >> So, do you > >>> have any suggestions about how to deal with these issues? > >> There is a oracle module in ser - you could port it to openser ;-) > >> > >> regards > >> klaus > >> > >>> Another issue that came up is the number of connections > >> from Openser > >>> to the database. In our case, listening to five interfaces, > >> with tcp > >>> disabled and children=5, we get 28 connections to DB which > >> is a great > >>> waste of resources. From those five interfaces, one is > >> receiving the > >>> bulk of traffic and the rest receive minimal traffic. Since > >> each child > >>> has its own connection, then what is the purpose of > >> connection pooling? > >>> How difficult would it be to have a common connection > pool for all > >>> children? > >>> > >>> sorry for the long email and thank you in advance for any answer. > >>> > >>> George > >>> > >>> > >>> > >>> > >>> > >>> Disclaimer > >>> The information in this e-mail and any attachments is > >> confidential. It is intended solely for the attention and > use of the > >> named addressee(s). If you are not the intended recipient, > or person > >> responsible for delivering this information to the intended > >> recipient, please notify the sender immediately. Unless > you are the > >> intended recipient or his/her representative you are not > authorized > >> to, and must not, read, copy, distribute, use or retain > this message > >> or any part of it. E-mail transmission cannot be guaranteed to be > >> secure or error-free as information could be intercepted, > corrupted, > >> lost, destroyed, arrive late or incomplete, or contain viruses. > >>> > >>> > >>> > >>> > >> > --------------------------------------------------------------------- > >> - > >>> -- > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >> > >> -- > >> Klaus Darilion > >> nic.at > >> > >> > > > -- > Klaus Darilion > nic.at > >

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[Users] openser+postgresql
by 赵俊杰 15 Mar '07

15 Mar '07

Hello,everyone I configure the openser with postgres.But I cann't start the openser .When I execute the command /usr/local/sbin/openser I occurs the following errors: 0(28102) ERROR: load_module: could not open module </usr/local/lib/openser/modules/postgres.so>: /usr/local/lib/openser/modules/postgres.so: undefined symbol: PQescapeStringConn 0(28102) parse error (42,13-14): failed to load module ERROR: bad config file (1 errors) 0(28102) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy And the file postgres.so exists and the path in the file openser.cfg is corrcet.Who can help me ,thank you! Junjie Zhao 2007-03-15

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[Serusers] ser_ctl - connection refused
by Terry Robert 15 Mar '07

15 Mar '07

Hi, I'm using ser_ctl as follows: ser_ctl -g -s http://localhost:2046 ps but get the following error: Traceback (most recent call last): File "/usr/local/bin/ser_ctl", line 19, in ? main(sys.argv) File "./serctl/main.py", line 173, in main return call(func, args, opts) File "./serctl/main.py", line 115, in call return apply(func, args, opts) File "./serctl/ctlctl.py", line 235, in ps ret = rpc.core_ps() File "./serctl/ctlrpc.py", line 195, in core_ps ps = self.ser.core.ps() File "/usr/lib/python2.3/xmlrpclib.py", line 1029, in __call__ return self.__send(self.__name, args) File "/usr/lib/python2.3/xmlrpclib.py", line 1316, in __request verbose=self.__verbose File "./serctl/serxmlrpc.py", line 132, in request self._http_request(uripath, body, host) File "./serctl/serxmlrpc.py", line 104, in _http_request self.conn.request("POST", uripath, body, headers) File "/usr/lib/python2.3/httplib.py", line 718, in request self._send_request(method, url, body, headers) File "/usr/lib/python2.3/httplib.py", line 739, in _send_request self.endheaders() File "/usr/lib/python2.3/httplib.py", line 712, in endheaders self._send_output() File "/usr/lib/python2.3/httplib.py", line 597, in _send_output self.send(msg) File "/usr/lib/python2.3/httplib.py", line 564, in send self.connect() File "/usr/lib/python2.3/httplib.py", line 548, in connect raise socket.error, msg socket.error: (111, 'Connection refused') In my ser.cfg, I have: # unix sockets and fifo creation parameters modparam("ctl", "binrpc", "unix:/tmp/ser_ctl") # default modparam("ctl", "binrpc", "udp:10.10.90.51:2046") modparam("ctl", "binrpc", "tcp:10.10.90.51:2046") modparam("ctl", "binrpc", "unixd:/tmp/unix_dgram") # unix datagram modparam("ctl", "binrpc", "unixs:/tmp/unix_stream") # unix stream modparam("ctl", "binrpc", "unix:/tmp/unix_default") # unix stream modparam("ctl", "binrpc", "/tmp/unix_test") # unix stream modparam("ctl", "binrpc", "10.10.90.51:3000") # udp modparam("ctl", "binrpc", "tcp:3012") # tcp any , port 3012 modparam("ctl", "binrpc", "udp:*:3012") # udp any , port 3012 modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") modparam("ctl", "mode", 0666) Any suggestions? Thanks.

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Re: [Users] Openser and Oracle
by Papadopoulos Georgios 15 Mar '07

15 Mar '07

Hi, Thank you for your offer. I have already been able to compile the oracle module from ser. However even this module will not solve the issue of so many connections to DB. best regards g. > > Hi... > > I'm using an modified oracle module from ser dev implementation, it > lacks raw_query, fetch_result and replace functions and needs more > testing as well. > > I Was waiting to have time to finish it before to send to the tracker, > but if you want to give it a try, i could send it to you. > > I havent tested it for memory leaks, so its possible to have. > > It's pretty stable in our test environment, but i never tested with a > high load. > > Best regards > 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.

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Re: [Users] Openser and Oracle
by Papadopoulos Georgios 15 Mar '07

15 Mar '07

Hi Christian, Our DB already has more than 100 connections from various other systems. Each connection takes about 5MB of memory. Since only 5 children are taking the heavy load, the rest 23 connections are mostly idle. And since we plan to have a second machine running Openser as a failover, there will be another 28 iddle connections. A single connection pool that would serve all children would be able to create connections on demand. At peak traffic it would be possible that the pool has as many connections as the number of children. But that would be rare (at least in our case). Basically, the amount of traffic would determine the amount of connections. What if each module opened the connection to the DB in mod_init() instead of child_init()? What implications could that have? Best regards George > Papadopoulos Georgios wrote: > ... > > Another issue that came up is the number of connections > from Openser to > > the database. In our case, listening to five interfaces, with tcp > > disabled and children=5, we get 28 connections to DB which > is a great > > waste of resources. From those five interfaces, one is > receiving the > > bulk of traffic and the rest receive minimal traffic. Since > each child > > has its own connection, then what is the purpose of > connection pooling? > > How difficult would it be to have a common connection pool > for all children? > > I don't think that having 28 DB connections is a waste of > resources, in > fact I think you will get better DB query performance using one DB > connection per openser worker process than using a DB > connection pool. > Most databases including Oracle DB are optimized for > concurrent access > using either a process or a thread pool, so why not take advantage of > that. And most DBs can easily handle at least 100 concurrent > DB connections. > > Christian > 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.

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Re: [Users] openser +ipv6
by Bernard Chan 15 Mar '07

15 Mar '07

於四，2007-03-15 於 09:58 +0800，赵俊杰提到： > Hello ,erveryone > > I configure the openser under ipv6.I add listen ipv6_adress in the > file openser.cfg,it cann't work.But under ipv4 it works perfectly.Who > can tell me how to configure ipv6 in openser.Thank you > very much! > OpenSER 1.2.0 can listen on IPv6 socket fine on both of our test systems. How do you write the listen directive? On my system, it looks like this: listen=udp:[2002:xxxx:xxxx::1:3]:15060 Regards, Bernard Chan.

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[Users] openser +ipv6
by 赵俊杰 15 Mar '07

15 Mar '07

Hello ,erveryone I configure the openser under ipv6.I add listen ipv6_adress in the file openser.cfg,it cann't work.But under ipv4 it works perfectly.Who can tell me how to configure ipv6 in openser.Thank you very much! 赵俊杰 2007-03-15

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[Users] Putting the OpenSER behind a load balancer which NATs
by Tim Madorma 14 Mar '07

14 Mar '07

Hi, I am going to put a pair of servers with OpenSER software on them behind a BIG IP load balancer which will use NAT to provide static IP addresses to each OpenSER server. In this case, when an incoming SIP Request is received, the destination IP address will differ from the host IP address defined in the Req URI (and To header). Is there any special configuration or a module that should be utilized to deal with this? thanks, Tim

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[Users] accounting radius and multileg calls
by Rosario Pingaro 14 Mar '07

14 Mar '07

Is there someone so kind to post an openser.cfg exemple on how to have multileg accounting working with acc_radius? Regards Rosario

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[Users] DBTEXT performance vs. MySQL?
by Daryl Sanders 14 Mar '07

14 Mar '07

Hi Everyone, I am wondering if anyone here has experience with DBTEXT and can comment on it's performance vs. MySQL. I'm involved in a project where it makes sense to use DBTEXT at least for the subscriber table, but I need to know what performance problems I might face once the system is loaded with subscribers. For example, how many calls per second or minute might I expect to see? All comments are welcome. Thanks. - Daryl

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[Users] differ between wrong password and stale nonce
by Klaus Darilion 14 Mar '07

14 Mar '07

Hi! I want to log failed authentication requests due to a wrong password (wrong response), but I do not want to log failed authentication requests due to a stale nonce. Is there a way to differ this 2 situations? MAybe are there different return values from proxy_authorize? regards klaus -- Klaus Darilion nic.at

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[Users] SIP supplementary service
by 김형선 14 Mar '07

14 Mar '07

Hi. I saw the example script of openser.cfg for 'call forwarding'. Are other SIP supplementary services possible with OpenSER, by script? (call parking, call pickup, hold, conference, etc...) If it is possible, how can I get these script files? thanks 새로운 기부문화의 씨앗, 해피빈

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[Serusers] Getting core dump for ser-0.8.14
by Gaurav Kulshreshtha 14 Mar '07

14 Mar '07

All, Is it possible to get core dump for ser-0.8.14 processes ? Thanks Gaurav

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Re: [Users] ACK ACK ACK
by Mark Kent 14 Mar '07

14 Mar '07

>> I guess you are forwarding the INVITE to the gateway using t_relay. >> Thus, you forward statefully by creating a transaction inside openser's >> tm module. Thus, for the transaction to complete, the tm module also has >> to see the ACK. Thus, you have to t_relay the ACK too. Thanks Klaus, that's what I've been doing. The thing that prompted me to email was a Grandstream phone that about 80% of the time will cause openser1.1.0 to go in a loop, blasting the ACK around internally, coming in the top of route[0] and subsequently being t_relayed() until I get too many hops. This is all internal, no packets are generated on the ethernet interface. I've looked at the ACK the Grandstream sends back and compared it to the ACK I get from a Linksys and they are almost identical. The only difference is the Linksys puts the port number on the ACK and To: like this: ACK sip:18005551212@openser.example.com:5060 SIP/2.0 To: <sip:18005551212@openser.example.com:5060>;tag=foo-9235 while the Grandstream doesn't mention the port: ACK sip:18005551212@openser.example.com SIP/2.0 To: <sip:18005551212@openser.example.com>;tag=foo-055c Even if the solution is to upgrade to 1.2.0 (which I'm going to do), if this was a known problem then I'ld appreciate hearing that. Thanks, -mark

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[Users] Openser and Oracle
by Papadopoulos Georgios 14 Mar '07

14 Mar '07

Hello, First of all congratulations to everyone involved in the new release. I haven't been able to get my hands on it yet, but just reading about it makes me very excited. Great job! I would like to address the issue of Openser and Oracle working together. We are currently using Openser with a local MySQL. Our main database is Oracle and we are just copying data to MySQL so that Openser can work. This is a little difficult to maintain so I thought I would try to make Openser use directly our Oracle. Of course I ran into a number of issues. 1. modules/acc and unixodbc. The acc_db_request() was not working because inserting a string in a date column does not work with Oracle. So, I had to change functions time2str() and time2odbc() in order to make this work. Also had to change acc.c to treat column time as DB_TIME instead of DB_STR (I think this could be treated as a bug). 2. modules/lcr. The query uses char_length() and rand() which I had to replace with lengthc() and dbms_random.value. Should these be modules params? 3. modules/avpops. All issues were resolved by config changes and replacing avp_db_load() with avp_db_query(). 4. modules/auth_db, alias_db, uri_db, group. Since MySQL is case insensitive and Oracle is not, I made changes in the code to use raw_query() instead of query(). All queries have to be in the form "select ... from ... where username=upper(...)" I would be happy to provide patches and help in any way in order to make this migration easier in the future. However I am not sure whether my changes are general enough for everybody to use. So, do you have any suggestions about how to deal with these issues? Another issue that came up is the number of connections from Openser to the database. In our case, listening to five interfaces, with tcp disabled and children=5, we get 28 connections to DB which is a great waste of resources. From those five interfaces, one is receiving the bulk of traffic and the rest receive minimal traffic. Since each child has its own connection, then what is the purpose of connection pooling? How difficult would it be to have a common connection pool for all children? sorry for the long email and thank you in advance for any answer. George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

4 3

[Serusers] RFC: call forwarding
by Nils Ohlmeier 14 Mar '07

14 Mar '07

Hello, as call forwarding was a frequently requested feature for the new => 2.0 SER configuration I recently added this feature to the ser.cfg in the HEAD branch repository. As this is a new "feature" I do not plan to add it to the 2.0 stable branch. But "backporting" should be easy ;-) So please give it a try and let me know your feedback. Thanks Nils

1 0

[Serusers] SIP tutorial has been updated
by Jiri Kuthan 14 Mar '07

14 Mar '07

The SIP tutorial by myself and Dorgham Sisalem has been updated and is available at http://www.iptel.org/sip. Comments to authors are most welcome. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Users] mi_xmlrpc on redhat
by Helge Waastad 14 Mar '07

14 Mar '07

Hi, does anyone know if there are any packages for the dependencies for the mi_xmlrpc module for RedHat/CentOS? (Im trying to get this working on my centOS dist) br

2 8

RE: [Users] Openser and Oracle
by Papadopoulos Georgios 14 Mar '07

14 Mar '07

It is uppercase in the code. I am using version 1.1.1 and it is in file lcr_mod.c, function do_load_gws. > -----Original Message----- > From: Henning Westerholt [mailto:henning.westerholt@1und1.de] > Sent: Wednesday, March 14, 2007 1:54 PM > To: users(a)openser.org > Cc: Papadopoulos Georgios > Subject: Re: [Users] Openser and Oracle > > On Wednesday 14 March 2007 11:25, Papadopoulos Georgios wrote: > > Hello, > > [..] > > 2. modules/lcr. The query uses char_length() and rand() > which I had to > > replace with lengthc() and dbms_random.value. Should these > be modules > > params? > > Hello Papadopoulos, > > i tried to look up the problem with the mentioned functions > above. I don't find a char_length() call in the lcr module. > What version of openser do you use? In what file are the > problematic calls? > > Best regards, > > Henning > 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.

2 1

RE: [Users] Openser and Oracle
by Papadopoulos Georgios 14 Mar '07

14 Mar '07

What do you mean by "SQL injection"? I know it is not a pretty solution (especially changing the code) but I cannot think of an alternative. I can look into SER's oracle module. But that would mean that in the where clauses all string values should be surounded with "upper". Not a pretty solution either but it would work I guess. I briefly saw that Openser 1.2 has string transformations. So could uppercase and lowercase be included there? I am picturing something like avp_db_load("$ruri{s.lowercase}", ...) Of course this would only solve the issue with avpops. In order to solve the issue with auth_db for example, wouldn't we need new functions? Any comments about the multiple connections to the DB? > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, March 14, 2007 12:59 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Openser and Oracle > > Papadopoulos Georgios wrote: > > Hello, > > > > First of all congratulations to everyone involved in the > new release. > > I haven't been able to get my hands on it yet, but just > reading about > > it makes me very excited. Great job! > > > > I would like to address the issue of Openser and Oracle working > > together. We are currently using Openser with a local > MySQL. Our main > > database is Oracle and we are just copying data to MySQL so that > > Openser can work. This is a little difficult to maintain so > I thought > > I would try to make Openser use directly our Oracle. Of > course I ran > > into a number of issues. > > 1. modules/acc and unixodbc. The acc_db_request() was not working > > because inserting a string in a date column does not work > with Oracle. > > So, I had to change functions time2str() and time2odbc() in > order to > > make this work. Also had to change acc.c to treat column time as > > DB_TIME instead of DB_STR (I think this could be treated as a bug). > > 2. modules/lcr. The query uses char_length() and rand() > which I had to > > replace with lengthc() and dbms_random.value. Should these > be modules > > params? > > Hi! > > I guess oracle allows the defintion of new function. Then you > could write the functions char_length() and rand() which > would be just wrappers to lengthc() and dbms_random.value. > > > 3. modules/avpops. All issues were resolved by config changes and > > replacing avp_db_load() with avp_db_query(). > > Be careful - raw queries are vulnerable to SQL injection!!! > > > 4. modules/auth_db, alias_db, uri_db, group. Since MySQL is case > > insensitive and Oracle is not, I made changes in the code to use > > raw_query() instead of query(). All queries have to be in the form > > "select ... from ... where username=upper(...)" > > Again: Be careful - raw queries are vulnerable to SQL injection!!! > > > > I would be happy to provide patches and help in any way in order to > > make this migration easier in the future. However I am not sure > > whether my changes are general enough for everybody to use. > So, do you > > have any suggestions about how to deal with these issues? > > There is a oracle module in ser - you could port it to openser ;-) > > regards > klaus > > > Another issue that came up is the number of connections > from Openser > > to the database. In our case, listening to five interfaces, > with tcp > > disabled and children=5, we get 28 connections to DB which > is a great > > waste of resources. From those five interfaces, one is > receiving the > > bulk of traffic and the rest receive minimal traffic. Since > each child > > has its own connection, then what is the purpose of > connection pooling? > > How difficult would it be to have a common connection pool for all > > children? > > > > sorry for the long email and thank you in advance for any answer. > > > > George > > > > > > > > > > > > Disclaimer > > The information in this e-mail and any attachments is > confidential. It is intended solely for the attention and use > of the named addressee(s). If you are not the intended > recipient, or person responsible for delivering this > information to the intended recipient, please notify the > sender immediately. Unless you are the intended recipient or > his/her representative you are not authorized to, and must > not, read, copy, distribute, use or retain this message or > any part of it. E-mail transmission cannot be guaranteed to > be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. > > > > > > > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > -- > Klaus Darilion > nic.at > >

2 1

[Users] dialog module configuration question
by Andy Pyles 14 Mar '07

14 Mar '07

Hello, I have a question on how to configure the dialog module ( 1.2.x from cvs yesterday ). With my config, ( attached) I can make calls and have verified that the acc module is working correctly. My question is, when I enable the dialog module, I can see that it is incrementing call count correctly, but when a bye is received, the dialog:active_dialogs statistic is never decremented. In the debug level 9 logs, ( also attached) I see this error after the 200OK is sent to the bye: 1(969) DBUG:dialog:unref_dlg: unref dlg 0xa7ce5a98 with 1 (delete=0)-> 1 Is this a case of one of the timers being set too short? by the way using a variable call length from well under a second ( using sipp ) to 20 second call doesnt' seem to make a difference . Thanks, Andy

5 29

[Users] OpenSER-1.2 Radius Auth
by TCB 14 Mar '07

14 Mar '07

Been trying to a fresh install of OpenSER-1.2 / Radius and i keep getting the following error on authentication , find attached . Googling has send me in circles i'm dizy. -- TC

2 1

[Users] Compiling new OpenSER 1.2.0 on Solaris 8 fails
by Peter Edwards 14 Mar '07

14 Mar '07

Hi, all. I have a Solaris 8 server which happily builds a nightly CVS build I retrieved from CVS on 16/12/06. When I heard that 1.2.0 had been released, I downloaded this tarball: http://www.openser.org/pub/openser/latest/src/openser-1.2.0-tls_src.tar.gz However, I'm having difficulties compiling it. To avoid a couple of symbol referencing errors, I added: -DDBG_QM_MALLOC \ and disabled: #-DF_MALLOC \ in the DEFS variable definition in Makefile.defs. I reached this by trial and error and by comparing the differences in the file from the working nightly build I can compile and the latest release tarball. However, I'm still getting: Undefined first referenced symbol in file warn cfg.tab.o ld: fatal: Symbol referencing errors. No output written to openser collect2: ld returned 1 exit status gmake: *** [openser] Error 1 Can anyone put me out of my misery? ... I'm not a C guru and am struggling a bit. Peter.

2 2

[Users] using regular expressions in switch case
by Jayesh Nambiar 14 Mar '07

14 Mar '07

Hi, First of all congratulations to all the contributors for releasing 1.2.0. I was just wondering if regular expressions can be used inside case " ----" syntax. Example: switch($ruri.user) { case "1[0-9]" : route(10); } Any help would be appreciated. w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

2 1

Fwd: [Users] NAT issue with devel version
by vallimamod abdullah 14 Mar '07

14 Mar '07

Sorry, forgot to put the list in copy :-(( -- Regards, -vma . ---------- Forwarded message ---------- From: vallimamod abdullah <vabdulla(a)gmail.com> Date: Mar 14, 2007 11:05 AM Subject: Re: [Users] NAT issue with devel version To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Hi Bogdan, On 3/10/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > Hi, > > could you detail a bit what is not working, as it is not clear from your > email ? Thank you for having taken time to look at my issue. I finally managed to solve it: It was not related to Openser but to my firewall who had some issues with NAT handling. -- Regards, -vma .

1 0

[Serusers] Environmental parameter
by chungyu 14 Mar '07

14 Mar '07

Dear all: Could tell me to have the environmental parameter can only get the username of from header and the domain of from header? ex: caller: john(a)iptel.org callee: mary(a)iptel2.org use $SIP_OUSER => mary $SIP_RURI => iptel2.org $xxx => john $xxx => iptel.org exec_msg("echo reguest from $XXX | mail -s 'request for you' xx(a)xx.xx.xx"); $xxx =>username of from header Thank you changyu 2007/03/14

1 0

[Users] ACK ACK ACK
by Mark Kent 14 Mar '07

14 Mar '07

Hello, It's been a little while since I last posted, but this will look familiar to those who remember: The scenario is this: Phone sends openser an invite, which tries to set things up with the PSTN gateway, which is unresponsive(*), and so it sends back to the Phone: SIP/2.0 503 Service not available and the Phone sends back an ACK. What should I do with that ACK? One possibility is to do nothing, another possibility is to t_relay() it. What suprised me was that when I do nothing then another "SIP/2.0 503 Service not available" message is automagically sent to the Phone. So, that can't be the right thing to do. Meanwhile, when I t_relay() it then things usually work. But this doesn't seem like the right thing to do, since the ACK is meant for the very box on which I'm doing the t_relay(). So, what's the story? Thanks, -mark (*)don't worry about why the PSTN gateway is unresponsive.

2 1

[Users] RTPproxy problem
by Joao Pereira 14 Mar '07

14 Mar '07

Hellos I whant to run RTPproxy separatly from my SIP router, so I tried to run it like this: [root@vox rtpproxy-0.3]# rtpproxy -s udp:xxx.xxx.xxx.xxx:22222 rtpproxy: explicit binding address has to be specified in UDP command mode [root@vox rtpproxy-0.3]# rtpproxy -l xxx.xxx.xxx.xxx: -s udp:xxx.xxx.xxx.xxx::22222 xxx.xxx.xxx.xxx: = my IP But when I run openSER, this happens 9(15400) WARNING: rtpp_test: support for RTP proxy has been disabled temporarily 11(15407) ERROR: send_rtpp_command: can't connect to RTP proxy I already tried with modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") and with modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock udp:xxx.xxx.xxx.xxx:22222") But it always says: send_rtpp_command: can't connect to RTP proxy Whats the problem? Im using this configuration in sip.conf: http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html Regards Joao Pereira

2 1

[Serusers] How to send ALIAS as caller ID
by Jai Rangi 14 Mar '07

14 Mar '07

Hello, I have working ser with all the db modules and lcr. I want to send useralias as caller id in place of user name. Say I have a ser account with the name testaccount, and an alias 15556667777 to the user testaccount. So if some one calls 15556667777 and the call will go to testaccount. Now if testaccount make an call then in the caller id I see testaccount. How can I send 15556667777 as caller ID when, testaccount make a PSTN call. I guess I can also use uri table and use the caller id from uri_user in place of username. But I dont know how can I do that. I tried this, modparam ("uri", uri_user_column", "uri_user") where the default is set for username. But That did not help. Thank you, -Jai

3 2

[Users] avp_db_query examples
by Klaus Darilion 13 Mar '07

13 Mar '07

Hi! With help of google I found http://www.openser.org/docs/avp_db_query.html which I could not find linked somewhere. IMO this would be a good idea. regards klaus -- Klaus Darilion nic.at

2 1

[Users] another svn question
by Klaus Darilion 13 Mar '07

13 Mar '07

Hi! I've checked out branch 1.2 using: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.2/ openser ... ... A openser/radius.h A openser/qvalue.c U openser Checked out revision 1836. Why has it checked out revision 1836? commit 1836 was Juha's LCR AVP changes which were applied to trunk. Taking a look at Makefile.defs I see that I've checked out 1.2, but why does svnversion report 1836? Maybe someone can shortly explain this. thanks klaus -- Klaus Darilion nic.at

4 3

[Users] OpenSER 1.2.0: no user password in database
by Franz Edler 13 Mar '07

13 Mar '07

Hi, I have got the following strange problem: after creating a user with openserctl add alice alice alice(a)netx.test database engine 'MYSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter=0 MySql password for user 'openser@localhost': new user 'alice' added root@sip:/# the password column in the database of user alice is empty, and I only can register without any password. The same happens with 'bob' etc... I am using plaintext passwords: modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") Any ideas what could have gone wrong? cheers Franz

2 2

[Users] Adding Users to 1.2.0
by Charles Austin 13 Mar '07

13 Mar '07

Hello, I installed 1.2 and am trying to add user accounts using the following command: openserctl add Bob xyzzy Bob(a)gigef1.net It let me add one user account with no problems. When I tried adding the second account i get the following error: ERROR 1062 (23000) at line 1: Duplicate entry ' ' for key 3 Any clues would be appreciated. Thanks, Charlie

2 3

[Users] Openser 1.2.0
by Danilo Vilela Dantas 13 Mar '07

13 Mar '07

Hi all, Is Openser 1.2.0. ready? Regard Danilo

4 3

[Users] MediaProxy and T.38 Fax
by Daryl Sanders 12 Mar '07

12 Mar '07

Has anyone been successful sending/receiving T.38 faxes with the latest version of MediaProxy? I have read that there were some issues with NAT previously. Is this still the case? Thanks! Daryl

3 2

[Users] sst bye issue
by Andy Pyles 12 Mar '07

12 Mar '07

Hi, I'm trying to understand how the SST handles the case of a UA that is off the network when the session timer expires. I have the following configuration: x-lite has session timer support, so using that to test the module. min session timer is set to 91 seconds. x-lite (149) - > openser --> x-lite ( 101) INVITE ---> -> INVITE <--100 trying 100 trying <----- <-180 ringing 180 ringing <-- <-200 ok 200 ok <--- ack ---> -->ack Call is setup at this point. I then remove the power supply from 101. some seconds later we see this: INVITE -------> --> INVITE 100 trying <------ ...... ( no response from 101) <--- 408 timeout BYE ----------> .... Ok at this point openser never sends a 200 ok to the BYE. >From what I can see in the SST module, that it deletes the dialog.in memory, but not sure how that will help here. --- debug logs -- where the FIRST bye comes in from 149: 0(5526) SIP Request: 0(5526) method: <BYE> 0(5526) uri: <sip:101@192.168.0.104:5061> 0(5526) version: <SIP/2.0> 0(5526) parse_headers: flags=2 0(5526) Found param type 232, <branch> = <z9hG4bK-d87543-8f727a53ca51673e-1--d87543->; state=6 0(5526) Found param type 235, <rport> = <n/a>; state=17 0(5526) end of header reached, state=5 0(5526) parse_headers: Via found, flags=2 0(5526) parse_headers: this is the first via 0(5526) After parse_msg... 0(5526) preparing to run routing scripts... 0(5526) parse_headers: flags=100 0(5526) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(5526) parse_headers: flags=10 0(5526) DEBUG: add_param: tag=faea2a83cd5fd6c1 0(5526) DEBUG:parse_to:end of header reached, state=29 0(5526) DBUG:parse_to: display={"101"}, ruri={sip:101@192.168.0.101} 0(5526) DEBUG: get_hdr_field: <To> [51]; uri=[sip:101@192.168.0.101] 0(5526) DEBUG: to body ["101"<sip:101@192.168.0.101>] 0(5526) DEBUG: add_param: tag=5077431e 0(5526) DEBUG:parse_to:end of header reached, state=29 0(5526) DBUG:parse_to: display={"149"}, ruri={sip:149@192.168.0.101} 0(5526) parse_headers: flags=200 0(5526) is_preloaded: No 0(5526) grep_sock_info - checking if host==us: 13==13 && [192.168.0.104] == [192.168.0.101] 0(5526) grep_sock_info - checking if port 5060 matches port 5061 0(5526) grep_sock_info - checking if host==us: 13==13 && [192.168.0.104] == [192.168.0.101] 0(5526) grep_sock_info - checking if port 5060 matches port 5061 0(5526) DEBUG:check_self: host != me 0(5526) grep_sock_info - checking if host==us: 13==13 && [192.168.0.101] == [192.168.0.101] 0(5526) grep_sock_info - checking if port 5060 matches port 5060 0(5526) after_loose: Topmost route URI: 'sip:192.168.0.101;lr;ftag=5077431e;did=5ff.8e4f3775' is me 0(5526) parse_headers: flags=200 0(5526) get_hdr_field: cseq <CSeq>: <3> <BYE> 0(5526) DEBUG: get_hdr_body : content_length=0 0(5526) found end of header 0(5526) find_next_route: No next Route HF found 0(5526) after_loose: No next URI found 0(5526) DBG:rr:run_rr_callbacks: callback id 0 entered with <lr;ftag=5077431e;did=5ff.8e4f3775> 0(5526) DEBUG:dialog:dlg_onroute: route param is '5ff.8e4f3775' (len=12) 0(5526) DEBUG:dialog:lookup_dlg: dialog id=1467217128 found on entry 4085 0(5526) DEBUG:dialog:run_create_callbacks: dialog=0xb5c65280, type=16 0(5526) DEBUG:sst_handlers.c:sst_dialog_terminate_CB:403: Terminating DID 0xb5c65280 session 0(5526) DEBUG:sst_handlers.c:sst_dialog_terminate_CB:410: Freeing the sst_info_t from dialog 0xb5c65280 0(5526) DBUG:dialog:unref_dlg: unref dlg 0xb5c65280 with 2 (delete=1)-> 0 0(5526) DBUG:dialog:destroy_dlg: destroing dialog 0xb5c65280 0(5526) loose_route() succeeded - M=BYE RURI=sip:101@192.168.0.104:5061 F=sip:149@192.168.0.101 T=sip:101@192.168.0.101 IP=192.168.0.102 ID=NTUzNzA5NWNjYmM5YWQ2MjQxYzgzZDdkZTRlMmQwODk. 0(5526) ERROR:dialog:dlg_status: res->ri = 5 0(5526) comp_scriptvar: int 20 : 5 / 0 0(5526) DEBUG: t_newtran: msg id=8 , global msg id=7 , T on entrance=0xffffffff 0(5526) parse_headers: flags=ffffffffffffffff 0(5526) parse_headers: flags=78 0(5526) t_lookup_request: start searching: hash=10869, isACK=0 0(5526) DEBUG: RFC3261 transaction matching failed 0(5526) DEBUG: t_lookup_request: no transaction found 0(5526) DBG: trans=0xb5c67b00, callback type 1, id 1 entered 0(5526) DBG: trans=0xb5c67b00, callback type 1, id 0 entered 0(5526) parse_headers: flags=78 0(5526) DEBUG: mk_proxy: doing DNS lookup... 0(5526) check_via_address(192.168.0.102, 192.168.0.102, 0) 0(5526) DBG:check_against_rule_list: using list dns 0(5526) DEBUG:tm:set_timer: relative timeout is 500000 0(5526) DEBUG: add_to_tail_of_timer[4]: 0xb5c67c4c (61500000) 0(5526) DEBUG:tm:set_timer: relative timeout is 30 0(5526) DEBUG: add_to_tail_of_timer[0]: 0xb5c67c68 (91) 0(5526) DEBUG:tm:t_relay_to: new transaction fwd'ed 0(5526) DEBUG:tm:UNREF_UNSAFE: after is 0 0(5526) DEBUG:destroy_avp_list: destroying list (nil) 0(5526) receive_msg: cleaning up --- thanks, Andy

2 3

[Users] OpenSER dies when MySQL is unreachable???
by Daryl Sanders 12 Mar '07

12 Mar '07

Hi Everyone, I just installed OpenSER w/MySQL and noticed that whenever MySQL became unavailable I could no longer place calls. Is this the correct behavior? I thought OpenSER's in-memory database would still allow new calls to go through. I'm using db_mode 1 in my openser.cfg on SVN 1.2.0 version of OpenSER. I'd really like to have OpenSER be able to continue working in the event MySQL is unreachable temporarily, for example, if the MySQL server crashes and needs to be rebooted or repaired. How can I accomplish this? Thanks! Daryl

5 9

[Users] Info Permissions - Trusted IP
by Dan-Cristian Bogos 12 Mar '07

12 Mar '07

Guys, Is it possible that permission module doesn't allow requests from a host which is sending from a port other than 5060 (my case 5061), even if this host is in trusted ip table? All other trusted hosts are working fine, but are sending from 5060. Thxs in advance, Dan

1 0

[Serusers] Ser-0.9.6 and exec_dset
by Tobias Lindgren 12 Mar '07

12 Mar '07

Hi all, I'm using exec_dset to run a script which takes the Request-URI and returns a new Request-URI based on the number dialed. In ser.cfg I'm doing this: if (exec_dset("lookup.sh \"$SIP_RURI\"")) { # Route call } else { # Reply with a cause } Recently a customer started to use SIP-X, and his INVITES with Request-URIs looks like this: sip:1001@192.168.0.1:5060;sipx-noroute=Voicemail Ser responds with "500 Server Internal Error" to these INVITEs, which is due to that my script fails. If I run SER in the foreground with debugging I can see this: "sh: line 1: sipx-noroute=Voicemail: command not found" so clearly the ";" in the INVITE causes a problem. How do I correctly escape the SIP_RURI in my ser.cfg to prevent my script from failing? If I run my script from commandline with the request uri, everything is fine. Br, /Tobias

1 0

[Users] SIP Proxy configuration
by Michel Bensoussan 12 Mar '07

12 Mar '07

Hello I need help in configuration. Lets say I already have a SIP Server (anyone) and I use it to handle VoIP in my company. Now I want to use OpenSER as a SIP proxy between the UAs and the SIP Server. I want all SIP messages to pass via OpenSER and the OpenSER shall be able to reject an INVITE message in both direction (incoming call and outgoing call). That's all. All the rest shall be handled by the SIP Server (forking, call forwarding, PSTN Gateway connectivity, etc...). Actually I just need the OpenSER receiving the requests, adding itself on route and forwarding to the SIP Server for outgoing requests and to the UA for incoming requests. Is there an example of config file that can do that? Thanks. Regards, Michel.

1 0

[Users] dbtext DB_STRING test
by jh＠tutpro.com 12 Mar '07

12 Mar '07

Ovidiu Sas writes: > I'm not familiar with the dbtext module and therefore I don't have a > patch for it, just the lcr patch that I already posted in the > tracker. ovidiu, but my point was that it doesn't make sense to remove DB_STRING test from one module and leave it in some other modules. i still believe that the right approach is to fix dbtext module by adding support for DB_STRING test there. so you should remove the bug tracker item on lcr and move it to dbtext module. -- juha

2 1

[Users] name for branch 1.2.x
by Daniel-Constantin Mierla 12 Mar '07

12 Mar '07

Hello, so far, when a new major release a new branch was created with the first version in that branch: rel_1_0_0 - was for 1.0.x rel_1_1_0 - was for 1.1.x Some complained that it is not suggestive when minor release in the branch was done, like 1.0.1 or 1.1.1. So, let's try to find a way more convenient. So we will have release a.b.c, where c can be incremented several time, being like a patch update release for a.b.0. Proposed options: 1) a_b_0 - stick to current module 2) a_b_x - use x to signal that it is a changing number for same major release 3) a_b - ignore patch release number Other suggestions are welcome. Please feedback as soon as possible. The branch will be created in few hours. Cheers, Daniel

4 5

[Serusers] Question for incoming PSTN calls
by Giovanni Chirico 12 Mar '07

12 Mar '07

Hi, I have my subscribers registered with a specific domain (ie: 0123456789(a)company.com) in subscribers table. The subscribers are geographic numbers and can made PSTN calls (outgoing calls). The question is for incoming PSTN calls: the gateway (Cisco) transfer calls to SER as <mailto:0123456789@192.168.0.1> 0123456789(a)192.168.0.1 (the IP of SER Server) and receive the error Forbidden because the subscriber is registered as <mailto:0123456789@company.com> 0123456789(a)company.com. Can I replace @192.168.0.1 with @company.com or disabile this check? Can someone help me? Thanks in advance, Giò

1 0

[Users] RTPproxy just for inside calls
by Joao Pereira 12 Mar '07

12 Mar '07

Hello Im configuring a server with hundreds of users and Im not using RTPproxy in order to allow more calls / second. But my clients that are behind NAT need OpenSER to force the RTP to pass through the server.... how can I configure RTPproxy to be used only in the calls inside my domain? Thanks regards Joao Pereira

1 0

[Users] cancelling forked requests + forcing TLS use
by Andrew T Gin 12 Mar '07

12 Mar '07

Hi I have two questions. 1. Receiver(a)1.1.1.1 and Receiver(a)2.2.2.2 register with ser. Client sends an INVITE to ser for Receiver. ser forks these INVITES in parallel to both Receiver(a)1.1.1.1 and Receiver(a)2.2.2.2. As both Receivers are online, they both reply with 200 OK. SER then relays BOTH 200 OK's back to the Client. My question: Shouldn't SER send CANCEL to one of them, and relay only ONE 200 OK back to the client? How would I do this? It is not the client's responsibility to send a CANCEL to the SER to relay to one of the receivers, is this correct? 2. When I have tls_verify_client = 1 tls_require_client_certificate = 1 in my cfg file, ser still accepts tcp and udp connections, despite tls_require_client_certificate=1. I did a search, and found this way: if (proto != TLS) { sl_send_reply("403", "Forbidden"); exit; }; However SER is really still listening on UDP and TCP port 5060. Is there a way to configure SER such that it only listens and acknowledges TLS connections? Thanks Andrew

3 2

[Users] OpenSER's doxygen documentation available online
by Daniel-Constantin Mierla 12 Mar '07

12 Mar '07

Hello, ... although mainly intended for developers, it might be useful for everybody ... Doxygen-generated documentation for development version is now available for browse online at: http://www.openser.org/doxygen-devel/ It is a very good tool to check where a variable or function was defined or referenced, look at internal structures without downloading sources locally. Credits to Norman Brandinger for the config files of doxygen. Those files are now included in source tree, in 'doc/doxygen', so if you want to generate a local copy of such documentation, use: # doxygen doc/doxygen/openser-doxygen The outup is in 'doc' directory. you can tune it from doxygen config files. Cheers, Daniel

2 1

[Users] lcr module parameters
by jh＠tutpro.com 12 Mar '07

12 Mar '07

Daniel-Constantin Mierla writes: > deadline was till today. Please no changes in the SVN only bug fixes if > found in last minute. Not other updates to C code -- docs are always > welcome to improve. ok, i'll commit the changes to trunk after 1.2 has been released. -- juha

1 0

Re: [Users] Dispatcher and failover in 1.2.0
by jh＠tutpro.com 12 Mar '07

12 Mar '07

Daniel-Constantin Mierla writes: > > There is another hidden bomb inside the lcr module. > yes, know about that. Talked with Juha and there seems to be loat of > changes to me made in time for this release. It is his decision how far > to go. I changed to many modules, where I had good knowledge. > Unfortunately lcr is not one I played too much, so the best is to have > maintainer do it. For next release, these should be labeled as "to be > fixed". i made the changes, but was out of "office" for a few days and have not tested them. when is the deadline? -- juha

3 2

[Users] Help needed with mediaproxy and rewritehostport configuration
by Berry Bartels 12 Mar '07

12 Mar '07

Im trying to set up a SIP and RTP proxy server to provide sip trunks to other SIP servers and PABX. The trunk to the PSTN will be provided by an other SIP server. This server will only handle signaling and rtp streams from our server. To test the configuration Im using a softphone (eyebeam 1.5), a cisco 7912 with SIP IOS, both logged in on my own SIP server, and a mobile phone as a PSTN client. Above shown in a diagram. http://praatpaal.com/images/sipdiagram.jpg The following occurs: 1. When I call from the softphone (eyebeam) the the Cisco 7912 the call is established correct. As far as I can debug the clients are forced to use the mediaproxy running on our server. This also works the other way around. 2. When I call from PSTN to a SIP client, I can only hear audio from the PSTN on the SIP client. 3. When I call from a SIP client to the PSTN, I can only hear audio from the SIP client on the PSTN client. So, my conclusion is: When a call is setup from or to the PSTN network, audio is only working from the client the call is originated. Below is the configuration from our SIP server and The Mediaserver. Please help me out here. Let me know if you have any questions or need more information Kind regards, Berry Bartels RoutIT B.V. OPENSER.CFG listen=XXX.XXX.XXX.XXX alias=" XXX.XXX.XXX.XXX " alias="voipit.nl" alias="sip01.voipit.nl" alias="sipproxy01.voipit.nl" debug=3 fork=no log_stderror=yes check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) sip_warning=yes syn_branch=yes server_signature=yes port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:********@localhost/openser" # --- load modules mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "domain.so" loadmodule "nathelper.so" loadmodule "mediaproxy.so" loadmodule "xlog.so" loadmodule "acc.so" # --- module params # rr # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # mediaproxy modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "natping_interval", 0) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "sipping_from", "sip:ping@voipit.nl") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("usrloc","db_url","mysql://openser:********@localhost/openser") modparam("acc", "db_url", "mysql://openser:********@localhost/openser") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) # request routing logic route { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: ROUTE script\n"); # sanity checks if (!mf_process_maxfwd_header("10")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Too many hops\n"); sl_send_reply("483","Too Many Hops"); exit; } if (msg:len >= max_len) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Message too big\n"); sl_send_reply("513", "Message too big"); exit; } if(loose_route()) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Loose Route -> route[1]\n"); t_relay(); } if (!method=="INVITE") { record_route(); } lookup("aliases"); if (uri==myself) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration :: URI is myself\n"); if (method=="REGISTER") { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("voipit.nl", "subscriber")) { www_challenge("voipit.nl", "0"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration :: Failed\n"); exit; } xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration :: Success\n"); save("location"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Registration :: Location Saved\n"); exit; } if (is_method("INVITE")||is_method("ACK")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Invite\n"); if (lookup("location") && from_uri=~".*(a)voipit.nl") { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Lookup Location :: Found :: Internal Call\n"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Internal Call -> Route[1]\n"); route(1); exit; } else if (lookup("location") && from_uri!=~".*(a)voipit.nl") { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Lookup Location :: Found :: Inbound Call\n"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Inbound Call -> Route[1]\n"); route(1); exit; } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Lookup Location :: NOT Found :: Outbound Call\n"); # sl_send_reply("404","Not found :: Outbound Call"); xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Outbound Call -> Route[2]\n"); route(2); exit; } } if (is_method("SUBSCRIBE")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Subscribe :: Subscribed in Database\n"); exit; } append_hf("P-hint: usrloc applied\r\n"); } xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: End of script -> Route[1]\n"); route(1); } route[1] { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[1]\n"); if (is_method("INVITE")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[1] :: INVITE\n"); t_on_reply("1"); # t_on_failure("1"); use_media_proxy(); } else if (is_method("BYE|CANCEL")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[1] :: BYE OR CANCEL\n"); end_media_session(); } if (!t_relay()) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Route[1] :: T_RELAY error\n"); sl_reply_error(); end_media_session(); } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Route[1] :: T_RELAY\n"); } exit; } route[2] { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[2]\n"); if (is_method("INVITE")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[2] :: INVITE\n"); rewritehostport("YYY.YYY.YYY.YYY:5060"); t_on_reply("2"); use_media_proxy(); } else if (is_method("BYE|CANCEL")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: route[2] :: BYE OR CANCEL\n"); end_media_session(); } if (!t_relay()) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Route[2] :: T_RELAY error\n"); sl_reply_error(); end_media_session(); } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: Route[2] :: T_RELAY\n"); } exit; } onreply_route[1] { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[1]\n"); if (status =~ "(183)|(2[0-9][0-9])") { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[1] :: Status is 183 or 200-299\n"); if (client_nat_test("1")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[1] :: nat_test(1) :: TRUE\n"); fix_contact(); } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[1] :: nat_test(1) :: TRUE\n"); } use_media_proxy(); } } onreply_route[2] { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[2]\n"); if (client_nat_test("1")) { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[2] :: nat_test(1) :: TRUE\n"); fix_contact(); } else { xlog("L_NOTICE", "[$Ts]\t[$fu]\t[$tu]\t[$rm] :: onreply_route[2] :: nat_test(1) :: TRUE\n"); } use_media_proxy(); } MEDIAPROXY.INI [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock group = openser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = openser listen = none allow = Any proxyIP = XXX.XXX.XXX.XXX portRange = 60000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 180 forceClose = 0 [Accounting] accounting = none [Database] user = openser password = ******** host = 127.0.0.1 database = radius table = radacct [Radius] secret = secret server = localhost authport = 1812 acctport = 1813 dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/radius/dictionary, /usr/local/mediaproxy/dictionary retries = 2 timeout = 3

2 1

[Users] Temporarily suspend or disable subscriber ???
by Daryl Sanders 12 Mar '07

12 Mar '07

Hi Everyone, Does OpenSER have any mechanism to temporarily suspend or disable a subscriber? For example, if they don't pay their bill. If not, how are other people doing this? I know I could modify their username or password in the subscriber table, but that seems like a bit of a hack. Any ideas? Thanks! Daryl

3 2

Re: [Users] [user] radius_is_user_in problem.
by jh＠tutpro.com 12 Mar '07

12 Mar '07

Bodin Bruno writes: > Sending Access-Accept of id 127 to 127.0.0.1 port 32773 > Reply-Message = "Authorized" if openser receives this and still reports failure, i would suspect your radiusclient library, since the result testing code in the module is exactly same as in other radius modules. or perhaps your openser does no get the reply at all. -- juha

1 0

[Users] openserctl avp issues
by Ovidiu Sas 10 Mar '07

10 Mar '07

Hi, openserctl avp add is failing due to $AVP_MODIFIED_COLUMN which should be "last_modified" (see openserctl.sqlbase). Regards, Ovidiu Sas

2 1

[Users] OpenSER with PRACK
by konstantinos nikou 10 Mar '07

10 Mar '07

Dear all, Does the Open SER support the PRACK-UPDATE method and if yes in which version? Thanks in advance. Best regards, K.N. ____________________________________________________________________________________ Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121

2 1

[Users] Status of t_uac_dlg in 1.2.0
by Daryl Sanders 10 Mar '07

10 Mar '07

Hi Everyone, Does anyone know if t_uac_dlg will still be available in OpenSER 1.2.0., either from openserctl or xmlrpc? We'd like to use this function to send fake byes when billing records are left open, that is unless there is a better way to acomplish this. Thoughts? Regards, Daryl

2 1

[Users] NAT issue with devel version
by vallimamod abdullah 10 Mar '07

10 Mar '07

Hello, I am testing 1.2.0-pre8-notls version running with mediaproxy and I have noticed someting very strange: When a client is behind nat it never connects to the mediaproxy for the RTP exchange. I have checked the sip traces, the media IP translation is done correctly in the SDP body. My setup has already worked some time ago (with devel version) and I haven't made big changes since. Were there changes on the way nat is handled that I have missed ? I tried to set the "received_avp" and "nat_bflag" params but this don't help. If somebody have a hint for me, I would be glad to hear it. Thank you ! -- Regards, -vma .

2 1

[Users] Can the timers be changed in situ?
by Mahesh Paolini-Subramanya 09 Mar '07

09 Mar '07

For various (and quite horrible) reasons, I need to do the following a) Wait '2' seconds after I send an INVITE for *any* response b) Wait '4' seconds after I get a 100 response for a 18x (or higher) response c) Wait '60' seconds after the 18x for a 18x (or higher) response Sooo, can I change the timer *during* the transaction? i.e., something like fr_inv_timer = 2 if (response == 100) { reset fr_inv_timer to 4 seconds } elsif (response == 180) { reset fr_inv_timer to 60 } Or, is fr_inv_timer fixed once the transaction is setup? cheers -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

1 0

[Users] web interface
by Joao Pereira 09 Mar '07

09 Mar '07

Hello Does OpenSER has its ouwn web interface or we need to use serweb? And were can I download the latest serweb release? Thanks Joao Pereira

2 2

[Users] Wesip and openser: auto reconnect
by Helmut Kuper 09 Mar '07

09 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I found out, that when wesip runs and openser was restartet or openser was started after wesip wesip doesn't do an auto reconnect to openser when openser' unix socks are available to wesip again. I think it will give openser/wesip couple more robustnes when there is a kind of ato reconnect. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF8VOT4tZeNddg3dwRAqqJAJ9Wa/r6trK9tECVC0M+ivHYjGs8IQCeIjZS Ay+mT46+ZQDp0wj0D4J+h3I= =KBLt -----END PGP SIGNATURE-----

4 3

[Users] Dispatcher and failover in 1.2.0
by Helge Waastad 09 Mar '07

09 Mar '07

Hi, just wondering about dispatcher failover in 1.2.0 in 1.1.0 i had: modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("dispatcher", "force_dst", 0) modparam("dispatcher","flags", 2) modparam("dispatcher","use_default", 0) ds_select_dst("1", "0"); and in failure rout checking 408 timeout an using ds_mark_dst(); ds_next_dst() What I see is that the "down" proxy is not put as inactive ERROR:avp:search_first_avp: 0 ID or NULL NAME AVP! Any clues? br hw

3 4

[Users] CDRTool problem with multi-leg call accounting and radius
by Norman Brandinger 09 Mar '07

09 Mar '07

I've added a troubleshooting entry in the wiki: http://openser.org/dokuwiki/doku.php/troubleshooting:radius Hope this helps anyone else fighting this issue. Regards, Norm

2 1

[Users] Meeting at Von in San Jose
by Bogdan-Andrei Iancu 09 Mar '07

09 Mar '07

Hi everybody, I have a presentation about OpenSER in the Open Source panel at VoN Spring in San Jose, so I will be there from 18th to 23th of March. If there are people interesting in meeting for discussions around openser, VoIp, SIP related topics (beer not excluded :)), please drop me a private email. Regards, Bogdan

1 0

[Users] Handling 183 Session Progress Messages
by Hansi Lao 09 Mar '07

09 Mar '07

Hello All, Is there a proper way of handling 183 messages with SDP body generated from a PSTN gateway? I used this code snippet: if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]") if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); Unfortunately, the media SDP body included in the 183 Session Progress message still can't be heard on the calling party. Are there any workarounds on this? Thanks!

1 0

[Users] $avp(s:rpid) and another question...
by Helge Waastad 09 Mar '07

09 Mar '07

Hi, two small questions this time :-) 1.) From 1.2.x modules dok, default rpid_avp for auth is: modparam("auth", "rpid_avp", "$avp(s:rpid)") However setting this gives a failure: parse_avp_name: error - use type (s: or i:) in front of avp name Mar 8 21:33:46 proxy-02 /usr/sbin/openser[16128]: ERROR:auth:init_rpid_avp: invalid rpid AVP specs "$avp(s:rpid)" Mar 8 21:33:46 proxy-02 /usr/sbin/openser[16128]: ERROR:auth:mod_init: failed to process rpid AVPs Mar 8 21:33:46 proxy-02 /usr/sbin/openser[16128]: init_mod(): Error while initializing module auth The "old" way still works: modparam("auth","rpid_avp","s:rpid") 2.) Earlier I had modparam("tm","fr_inv_timer_avp","s:inv_timeout") and: avp_db_load("$tu",s:inv_timeout) [inv_timeout attribute in database] What would be the equivalent with the new avp configuration? br hw

2 7

[Users] sdp negotiation fails - no ACK - rtpproxy
by Benko 09 Mar '07

09 Mar '07

Hello! I'm struggling with a issue since several days, since i make no more efforts i'd like to ask you for help: I'm trying to attach a exchange 2007 server to openser, most of the features work(inbound call, refer) but i can't get a outbound call to work as exc doesn't seem to like the SDP in my "200 OK" answer, i get no ACK and no rtp-stream is sent from exchange. My setup looks like this: ________ _______ _____ ________ |asterisk|=======|openser|======|(NAT)|=======|exchange| |________| |_______| |_____| |________| 2x.x.x.107 2x.x.x.106 62.x.x.193 10.1.99.97 A rtpproxy is running on the openser too, exchange is talking tcp... In my openser-config i use the nathelper-module and rtpproxy to traverse nat for the clients. This setup is working for all phones i've tested so far, so imho it is useable. Exchange has a feature that allows dialouts to read voicemail to e.g. a mobile phone. It is using a standard INVITE dialog, in a local network without nat i've been succesfull to attach exchange to sipx and a phone registered at sipx and i guess it will also work with a local openser who doesn't use nat-traversal features. However, with the public openser where i have a NAT, exchange doesn't seem to accept my sdp-proposal, i get no ACK and the rtp stream is only sent by asterisk, not by exchange. Unfortunately exchange offers no valuable debugging information, so i have no idea what exactly it doesn't like about my answer - i did some trial and error with different nat-helper-functions, activated and deactivated them but so far it didn't make a difference(or the call didn't got through at all due to missing nat-traversal). I've uploaded wireshark-traces exported to text of the working local dialog with sipX(http://pastebin.ca/383516) and the non-working external dialog (http://pastebin.ca/383540) with openser, maybe someone can take a look at it and check if there's something wrong on my(=openser) side of the dialog - Thanks in advance! best regards Christian

2 2

[Users] Strange Problem involving PSTN call in
by Sam Lee 09 Mar '07

09 Mar '07

Hey Guys, I have had this very strange problem .. I have 2 PSTN gateways to allow for inbound and outbound calls. INBOUND calls from PSTN will reach one of these gateways (depending on the provider) and then directly routed to OPENSER. Gateways are cisco AS5350. The very strange thing is , on certain SIP-UAs that are being used outside (most of them ADSL users) , calls that come inbound via Gateway A can always ring the SIP-UA , calls coming inbound via Gateway B will not get a response. AND when i did a packet capture on the openser server , both gateways forwarded SIP requests inwards to the server and both are being processed . User Location is found and being routed. Both calls are being routed to the same UA , same port . So much for strange , anyone has any idea ? Regards, Sam

2 2

[Users] Openser 1.1.x avp question
by Mike O'Connor 09 Mar '07

09 Mar '07

Hi Guys Would this work ? (I have a working system and do not want to break it to test) avp_db_load("$ruri/username", "*"); if( is_avp_set("$avp(s:callfwdimmediate)")) { .... }; if( is_avp_set("$avp(s:callfwdbusy)")) { .... }; if( is_avp_set("$avp(s:callfwdnoanswer)")) { .... }; Thanks Mike

2 1

Re: [Users] WeSIP b2bua problem
by tele 09 Mar '07

09 Mar '07

Hi Gines! It's almost clear to me how to do, this what i have done and it works. now i need last clarification how forward the 200 OK from BYE request, i cannot get it working. for forwarding the 200 ok from the BYE in the doBye request i save downstreamLeg.setAttribute("byereq", req) and then in the doResponse i get the bye request and create the response from it, something like this but i'm wrong. if (resp.getStatus() == 200 && resp.getMethod().equalsIgnoreCase("BYE")) { SipServletRequest upstreamRequest = (SipServletRequest)upstreamLeg.getAttribute("byereq"); SipServletResponse upstreamResponse = upstreamRequest.createResponse(resp.getStatus(),resp.getReasonPhrase()); //Copy the content from the downstream response to the upstream response if (resp.getContentType() != null) { upstreamResponse.setContent(resp.getRawContent(), resp.getContentType()); } upstreamResponse.send(); } this is what i have done for working ACK. protected void doAck(SipServletRequest req) throws ServletException, IOException { //Retrieve the upstream request to respond it SipSession downstreamLeg = (SipSession)req.getSession().getApplicationSession().getAttribute("downstreamLeg"); SipServletResponse downstreamResponse = (SipServletResponse) downstreamLeg.getAttribute("200ok"); SipServletRequest ackRequest = downstreamResponse.createAck(); //Copy the content from the downstream response to the upstream response if (req.getContentType() != null) { ackRequest.setContent(req.getRawContent(), req.getContentType()); } ackRequest.send(); } protected void doBye(SipServletRequest req) throws ServletException, IOException { SipSession downstreamLeg = (SipSession)req.getSession().getApplicationSession().getAttribute("downstreamLeg"); SipServletRequest byeRequest = downstreamLeg.createRequest("BYE"); // Copy the content from the downstream response to the upstream response if (req.getContentType() != null) { byeRequest.setContent(req.getRawContent(), req.getContentType()); } byeRequest.send(); } protected void doResponse(SipServletResponse resp) throws ServletException, IOException { if (resp.getStatus() == 200 && resp.getMethod().equalsIgnoreCase("INVITE")) { SipSession downstreamLeg = (SipSession) resp.getSession().getApplicationSession().getAttribute("downstreamLeg"); downstreamLeg.setAttribute("200ok",resp); } //Retrieve the upstream request to respond it SipSession upstreamLeg = (SipSession)resp.getSession().getApplicationSession().getAttribute("upstreamLeg"); } On Mon, 2007-03-05 at 00:29 +0100, Ginés Gómez wrote: > The problem is that you didn't implement the doACK method. Implement > it following a similar technique as the one used in doResponse. Save > the 200OK response in the downstream session so you can retrieve it > when the ACK arrives from the upstream then generate ACK using > response.createACK() and copy the content using getContentType/ > setContentType geContent/setContent if required > > > Hope it helps > > Gines > > > Attached there is a .zip with all the log, trace and configuration. > > > > thanks for the support, > > > > :tele > > <wesip_test.zip> > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users >

2 10

[Serusers] INFO: Daily generated documentation
by Greger V. Teigre 09 Mar '07

09 Mar '07

Hi all, The daily generation of module documentation is now the following: * SER 2.0 aka Ottendorf module documentation can be found at http://www.iptel.org/views/moduledocs or in table from: http://www.iptel.org/views/tableofmodules * Module documentation for the current development (trunk) can be found at http://siprouter.onsip.org/doc/trunk/modules/ * Doxygen documentation for trunk can be found at http://siprouter.onsip.org/doc/doxygen/ All are generated nightly around 4am (CET). Thus, if you want to see documentation for modules added in the current development version, use http://siprouter.onsip.org/doc/trunk/modules/ All links on iptel.org have been updated, so these links can easily be found. g-)

1 0

[Users] accounting of final replies
by Henning Westerholt 08 Mar '07

08 Mar '07

Hello all, i need to do additional accounting for final replies with the acc module, especially distinguish local from forwarded replies. We patched 0.9.5 for this functionality. We add a flag to the message, that is set by tm, and read from the acc module and then added to the log message. Is this possible in 1.2 with the extra accounting functions and pseudo variables? Or is the approach described above still prefered? I know that there exist the t_local_replied() function in tm, and it is perhaps with that possible to log a message in the config file with add_log_request(). But it would be nice if this information could be included in the normal accounting log message. Thanks, Henning

2 1

[Users] Routing question
by Lucas Diaz 08 Mar '07

08 Mar '07

I have openser listening in 2 different interfaces, and it works great. My question is about users with different ip ranges: do they have to be able to see each other (same ip range or through a gateway), or openser must/can allow/route that communication? Thanx a lot. Lucas

2 1

[Users] Working on 3xx FWD
by Sam Lee 08 Mar '07

08 Mar '07

Hey guys, I have a current setup like this . PSTN user calls in to SIP boxes via PSTN Gateways Gateway will send the call to the OPENSER servers and in turn rings the SIP UA IF SIP UA tries to forward it out to the PSTN again , the problem i am getting now is NO ring back on the CALLER . (called party (PSTN) rings and has no problem starting conversation . Just that no ring back). I am currently making use of the UAC_Redirect in the openser server to route based on the 3xx move temporarily message. Is there a way to send a provisional message to the calling party PSTN gateway (originating GW) that it is trying / ringing so that the ORIGINATING GW will generate ring back to the caller ? Please let me know if you can't understand . THanks for your help guys! Regards, Sam

2 1

[Users] how to add user to database
by sukerry 08 Mar '07

08 Mar '07

Hi, I have installed openser 1.2.0 in my federo, however I do not know how to use a tool to add users to database. Any hint will be appreciated. Thanks kerry

2 1

[Serusers] cancelling forked requests + forcing TLS use
by Andrew T Gin 08 Mar '07

08 Mar '07

Hi I have two questions. 1. Receiver(a)1.1.1.1 and Receiver(a)2.2.2.2 register with ser. Client sends an INVITE to ser for Receiver. ser forks these INVITES in parallel to both Receiver(a)1.1.1.1 and Receiver(a)2.2.2.2. As both Receivers are online, they both reply with 200 OK. SER then relays BOTH 200 OK's back to the Client. My question: Shouldn't SER send CANCEL to one of the receivers, and relay only ONE 200 OK back to the client? How would I do this? It is not the client's responsibility to send a CANCEL to the SER to relay to one of the receivers, is this correct? This is because the client only sent one invite, therefore it should only receive one OK. ie should SER 'shield' the client from the knowledge of the existence of multiple contact addresses? 2. When I have tls_verify_client = 1 tls_require_client_certificate = 1 in my cfg file, ser still accepts tcp and udp connections, despite tls_require_client_certificate=1. I did a search, and found this way: if (proto != TLS) { sl_send_reply("403", "Forbidden"); exit; }; However SER is really still listening on UDP and TCP port 5060. Is there a way to configure SER such that it only listens and acknowledges TLS connections? Thanks Andrew

2 1

[Users] CLI and GUI
by Michel Bensoussan 08 Mar '07

08 Mar '07

Hello Is there is CLI (Command Line Interface) that enable a user to read or eventually to change parameters at run time in OpenSER? Is there a GUI (web, php,...) that enable configuration? Thanks. Regards, Michel.

2 1

[Serusers] Prague User Group Meeting Agenda
by Greger V. Teigre 08 Mar '07

08 Mar '07

Hi, Who's in charge of the agenda in Prague? (Or do we have somebody in charge? ;-) The presenters need to be allowed time to prepare for presentations etc. And we should decide on the length and the format of the meeting. I hope we can avoid a "death-by-powerpoint" (yes, I include openoffice in this category) style and rather have short intros and discussions :-) g-)

2 3

[Users] pdt and IPv6
by Bernard Chan 08 Mar '07

08 Mar '07

Dear All, I am experimenting with OpenSER on an IPv6 environment. The pdt module appears not to support using an IPv6 address as destination domain (I guess that is true of source domain too, but I do not need that anyway). I tried to append a pdt row to MySQL using the documented OpenSER FIFO interface. The message I sent to the FIFO is similar to :pdt_add:openser_tmp_reply voip.mycompany.com 9 [::ffff:abcd:1234] However, when the row is created in MySQL pdt, the [:: is apparently missing. I guess that is because the FIFO parser mistook that as the FIFO :-separated message header or something similar? Testing with an IPv4 address instead as the destination domain, the row can be created properly, and prefix2domain() worked flawlessly in an IPv4 environment. I would like to use IPv4-mapped IPv6 address to forward to an Asterisk PABX over SIP+IPv4 as Asterisk has no official IPv6 release yet (this is another question, whether OpenSER will handle v4-mapped address properly). This is what I would like to test by setting this pdt rule. I'm using OpenSER 1.2.0-dev19-notls on i386/linux, by the way. Thanks for any invaluable insights. Regards, Bernard Chan.

2 4

[Serusers] SEMS 0.10.0 rc2 released
by Stefan Sayer 07 Mar '07

07 Mar '07

Hi, We are happy to announce the release of SEMS 0.10.0 rc2. This release has a number of bugfixes and stability improvements, but also some interesting new application and component modules like PIN collect for scalable conference systems, a new flexible way to script SEMS in Python, SIP registrar client and UACi authentication, and a new method to handle jitter. Check it out at http://iptel.org/sems . From the Changelog - new Adaptive jitter buffer as alternative playout method Contributed by Andriy Pylypenko/Sippy Software - new PIN collect application with transfer to e.g. separate conference bridge - new SIP registrar client for registration at a SIP registrar - new UAC authentication component - new faster announcement application with memory caching for audio files - new pre call announcement method using REFER - stats server can be used for monitoring custom modules/applications - session specific parameters by default taken from unified session parameters header - signature configurable - install and make system updated - added documentation - some security bugfixes (namely fixing possible buffer overflows) - ...and a lot of other bug fixes Thanks go to Andriy and Maxim for the adaptive jitter buffer, Juha, Alexandr, Alex and all the others for bug reports and contributions. Regards Stefan for the semsdev team -- Stefan Sayer Media Services Development stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

1 0

[Serusers] SIP SER
by Goh, Tee-Yong 07 Mar '07

07 Mar '07

Dear all, Would appreciate your advice from the ser-0.9.6 program, if save.c is the program in the SER that reply the 200 OK message for the REGISTER sends from the client? Thanks. Regards, Howard This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer

3 2

[Users] [user] radius_is_user_in problem.
by Bodin Bruno 07 Mar '07

07 Mar '07

Hello, I try to use this code : if (!radius_is_user_in("From", "voip")) { sl_send_reply("403", "Forbidden"); exit; }; Radius server return this : Reply-Message = "Authorized" but in openser in debug mode there is : 0(12528) radius_is_user_in(): Failure Thank for your help.

2 1

[Users] RE: Openser and Sip provider
by Gene Willingham 07 Mar '07

07 Mar '07

It is possible to have the SIP provider route calls to an IP address. This will alleviate the need to register with the provider. Using a Linksys device may not be the answer either. The Linksys device can register with openser, but there are limitations on the handling of inbound calls. The linksys device assumes that the TO Header is the account name. There is a way to route the inbound DNID using the dial plan. You can contact Linksys support, they should be able to tell you. I would if I remembered how. For asterisk we modified the code (very simple change) to allow Openser to route calls to asterisk and still use the users peer definition. The change is a very small change 3 lines of code, that works with both 1.2 and 1.4 branch of Asterisk. We can share this with you. Gene Willingham Telasip -----Original Message----- Message: 1 Date: Tue, 6 Mar 2007 14:32:56 -0600 From: "Scott Yagel" <syagel(a)packetcall.net> Subject: [Users] Openser and SIP provider To: <users(a)openser.org> Message-ID: <000901c7602e$9fefbf40$4b67a8c0@syagel> Content-Type: text/plain; charset="us-ascii" Hello, Am I correct in my assumption that I can't directly connect Openser to a ITSP due to Openser not providing registration to the ITSP? This seems to be what I read in the Openser documentation. This being the case, I have a Linksys SPA9000 pbx that I can use as a gateway to the ITSP, but attempts to route 10-digit numbers to the pbx get a response of 403 Forbidden from the pbx (the pbx has registered to the Openser OK). I see that many use an asterisk for a gateway, how do they get around this problem with the asterisk? Thanks, Scott Yagel PacketCall, Inc. syagel(a)packetcall.net

2 1

[Serusers] Setup call forward by phone keyboard
by inge 07 Mar '07

07 Mar '07

Hi all, Is there a way to setup the call forward using the phone ? For example, proxy handle uri *73*xxxxxxxxxx#. Next is it possible to store "xxxxxxxxxx" into usr_preferences table like ? Thanks for your support

1 0

[Users] WeSIP for commercial use
by Helmut Kuper 07 Mar '07

07 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I remember, that WeSIP isn't free for commercial use. So I does anyone knows anything about pricing model? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF7Xqt4tZeNddg3dwRAi8FAJ9RZBjIfkmgiLtPfd14214AI6hllQCgvEP0 8ykyRmMA0EsGOj+1h0tiznE= =oHku -----END PGP SIGNATURE-----

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Re: [Users] imc
by Daniel-Constantin Mierla 07 Mar '07

07 Mar '07

Hello, On 03/05/07 19:25, Henri Keski-Sikkilä wrote: >> When user exit room only admin gets message "The user [user] has left >> the room" others doesn't know anything that user has left room... >> > Could you test with the latest CVS version? If you get the same, please > capture the sip traffic on openser box and send it to me for this case > ngrep -d any -qt port 5060 > > Cheers, > Daniel > > Hello, > > It works now, as you said. great. I will close the issue on the tracker > I have worked with sip and used openser 1.5 year now. I think openser is one of the best open source project that I know. It's high quality programming project. If there are some c programming/testing tasks I could do, it would be nice :) You could send me e-mail, if I could help the openser project in any way. > Coding and testing are very important for open source projects. If you feel like needing an extension, or improving existing ones, you are more than welcome. Also, if you think that a adjacent tool will make life easier, you can start develop it and make it available (e.g., scripts to get CDRs out of acc table ...). At this moment I'm busy with the new release, but afterwards, maybe would will publish a pool of "nice-to-have" stuff, and users can come and pick up one to implement. Cheers, Daniel > Thanks > Henri > > >

1 0

[Users] No rtpproxy when clients are behind the same nat
by Rafael J. Risco G.V. 06 Mar '07

06 Mar '07

Hi does anyone has a working example for this escenario? thanks rafael

6 6

[Users] WeSIP:Click2Call
by Helmut Kuper 06 Mar '07

06 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, Now that WeSIP is up and running and Click2Call application is installed and runs as well in WeSIP I tried to use Click2Call. The Website with Click2Call application is loaded and I have origin, password and domain filled out with some random data. I click on "Click2Call" button and ... nothing happens except that an error occurs in my browser's java console window. The error message is "DWRUtil is not defined". A click on it leads me to line 231 of the web sourcecode: var src = DWRUtil.getValue("src"); in function validate(). This is the same in both IE7 and Firefox 2.0.0.2 Nothing is sent to openser and in wesip logs u can found no actions, so I think even wesip does not get any information from the Click2Call webpage, which means that this must be a local error in the webpage's javascript code. What can I do to solve this? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF7Ym84tZeNddg3dwRArF9AJwNXRABje3g60xys3Y+F+SxKaOGKQCePoow OhZyOTZhpu+aj9Uan0sU+Yw= =d3r+ -----END PGP SIGNATURE-----

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