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[Users] OpenSER + Asterisk - Music On Hold
by Edoardo Serra 16 Mar '07

16 Mar '07

Hi all, I have an OpenSER server in front of serveral Asterisk acting as a load balancer and registrar server. We're offering both, inbound and outbound call services. When an outbound call is made, OpenSER, through the dispatcher module, choose an Asterisk server to handle the media of the call. When an inbound call is received (by a PSTN GW interconnected with one of the Asterisks), Asterisk calls SIP/username@openser. Media flows directly from user to Asterisks without using RTPProxy as every Asterisk server has got a public IP Address.. I have the following problem with MOH. If a user tries to put on hold an outbound call (placed by him) everything is OK, Asterisk start playing MOH and stops when the user wants to stop it. But, if a user wants to put on hold an inbound call (a call just answered), as soon as it press the hold button another call to the caller is originated and the first call is not put on hold by the Asterisk I guess the problem is that, in this case, the asterisk doesn't recognise the INVITE as a re-INVITE and originate a new call instead of putting the other on hold. Do you have any idea on how to solve the problem ? Every suggestion is appreciated. Regards Edoardo Serra

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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

1 0

[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-----

3 2

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-----

2 1

[Users] Problems when branching multiple legs to asterisk
by Ardjan Zwartjes 06 Mar '07

06 Mar '07

Hello list, Can anybody help me with the following problem: I have a network with a machine running OpenSER (1.1.0) and a machine running Asterisk (1.2.7). When a call is made to a certain number on the OpenSER machine I use a append_branch to fork the call to two phones, this works fine. The problem arises when I want to fork the call to two extensions on the asterisk machine. Only one of the extensions will actually start ringing. Using ethereal I found out that OpenSER sends two invites to the asterisk machine, asterisk returns a "100 Trying" on both invites, but after that Asterisk seems to forget about the second call. The first Invite is processed correctly. I suspect that the problem lies in the fact that asterisk receives two calls with the same Call-ID and just drops one of them. This behaviour is understandable since call-ids are supposed to be unique (as far as I know). Sending two invites with the same call-id on the otherhand also is the logical thing to do from OpenSERs point of view, but clearly this isn't "compatible". So my question is: Is there some way to let OpenSER succesfully branch a single call to two extensions on the same Asterisk? I hope somebody knows of a way how to fix this. Thanks in advance. Kind regards, Ardjan Zwartjes.

2 2

Re: [Users] TLS and dead tcp child
by Bogdan-Andrei Iancu 06 Mar '07

06 Mar '07

Hi Paulo,Hi Ohad, could you please send me the output of >\ openssl ciphers from your machines? regards, bogdan Paulo Angonese wrote: > My version of krb5-libs is krb5-libs-1.4.1-5. The newest for FC4... > > > Paulo Angonese escreveu: >> Bogdan, here is the bt: >> >> $ gdb /usr/local/sbin/openser core.9999 >> >> .... >> >> #0 0x00c63b87 in strchr () from /lib/libc.so.6 >> #1 0x0033dc53 in krb5_kt_resolve () from /usr/lib/libkrb5.so.3 >> #2 0x003cf5a0 in kssl_keytab_is_available () from /lib/libssl.so.5 >> #3 0x003bbc4c in ssl3_choose_cipher () from /lib/libssl.so.5 >> #4 0x003b73c0 in ssl3_accept () from /lib/libssl.so.5 >> #5 0x003c4eba in SSL_accept () from /lib/libssl.so.5 >> #6 0x080e1dc9 in tls_accept (c=0xb616b490) at tls/tls_server.c:236 >> #7 0x080e35cc in tls_fix_read_conn (c=0xb616b490) at >> tls/tls_server.c:872 >> #8 0x0809c428 in tcp_read_req (con=0xb616b490, >> bytes_read=0xbfad8448) at tcp_read.c:411 >> #9 0x0809c7a7 in handle_io (fm=0x814e1b0, idx=-1) at tcp_read.c:772 >> #10 0x0809dbf3 in io_wait_loop_epoll (h=0x8113020, t=Variable "t" is >> not available. >> ) at io_wait.h:722 >> #11 0x0809df67 in tcp_receive_loop (unix_sock=17) at tcp_read.c:893 >> #12 0x08093969 in tcp_init_children (chd_rank=0x810dcd4) at >> tcp_main.c:1745 >> #13 0x0806ce1e in main_loop () at main.c:1009 >> #14 0x0806e8cb in main (argc=1, argv=0xbfad86f4) at main.c:1477 >> >> >> >> >> Paulo Angonese escreveu: >>> Hi Bogdan, >>> >>> I´m sorry for my dumbness, but the backtrace log shows only this: >>> >>> [root@sx83 /]# gdb /usr/local/sbin/openser >>> GNU gdb Red Hat Linux (6.3.0.0-1.84rh) >>> Copyright 2004 Free Software Foundation, Inc. >>> GDB is free software, covered by the GNU General Public License, and >>> you are >>> welcome to change it and/or distribute copies of it under certain >>> conditions. >>> Type "show copying" to see the conditions. >>> There is absolutely no warranty for GDB. Type "show warranty" for >>> details. >>> This GDB was configured as "i386-redhat-linux-gnu"...Using host >>> libthread_db library "/lib/libthread_db.so.1". >>> >>> (gdb) run >>> Starting program: /usr/local/sbin/openser >>> Reading symbols from shared object read from target memory...done. >>> Loaded system supplied DSO at 0x6e9000 >>> 0(2634) WARNING: fix_socket_list: could not rev. resolve 10.128.63.183 >>> 0(2634) WARNING: fix_socket_list: could not rev. resolve 10.128.63.183 >>> Listening on >>> udp: 10.128.63.183 [10.128.63.183]:5060 >>> tls: 10.128.63.183 [10.128.63.183]:5061 >>> Aliases: >>> >>> 0(2634) TLS: Client verification activated. Client certificates are >>> mandatory. >>> 0(2634) TLS: Server verification activated. >>> Detaching after fork from child process 2637. >>> >>> Program exited normally. >>> (gdb) 11(2669) ERROR: receive_fd: EOF on 14 >>> >>> >>> The binaries are available in >>> ftp://ftp.procergs.com.br/pub/procergs/openser/openser-binaries.tgz >>> >>> SSH access is harder. This machine is in the intranet. >>> >>> Thanks >>> >>> Paulo >>> >>> Bogdan-Andrei Iancu escreveu: >>>> Hi Paulo, >>>> >>>> the core files without the binaries ( openser+modules (so) ) ar not >>>> usable. Let us first try the backtrace - could you send me the >>>> backtraces? Oterwise, please upload the binaries for download also. >>>> >>>> there is a simple way, if you could provide access to your machine >>>> (using ssh keys) and can directly inspect the core files. >>>> >>>> regards, >>>> bogdan >>>> >>>> Paulo Angonese wrote: >>>>> Ok. >>>>> >>>>> The core (two files was generated) and the full log (debug=9) are >>>>> in ftp://ftp.procergs.com.br/pub/procergs/openser >>>>> >>>>> >>>>> >>>>> Bogdan-Andrei Iancu escreveu: >>>>>> Hi Paulo, >>>>>> >>>>>> some suggestion :) - could you provide full log (debug) and >>>>>> core+binaries (for download) ? Or backtrace? >>>>>> >>>>>> regards, >>>>>> bogdan >>>>>> >>>>>> Paulo Angonese wrote: >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I have the same problem. >>>>>>> When the SNOM ( 360 SIP 6.5.2 ) sends the REGISTER using TLS >>>>>>> OpenSER crashes. >>>>>>> I´m using >>>>>>> OpenSer (1.1.1-tls (i386/linux)), >>>>>>> openssl-0.9.7f-7.10 >>>>>>> openssl-devel-0.9.7f-7.10 >>>>>>> Fedora 2.6.11-1.1369_FC4 >>>>>>> >>>>>>> Any help? >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> Paulo Angonese >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> [Users] CVS 31.03, TLS and dead tcp child >>>>>>> Bogdan-Andrei Iancu bogdan at voice-system.ro >>>>>>> Tue Oct 31 17:13:43 CET 2006 >>>>>>> >>>>>>> * Previous message: [Users] CVS 31.03, TLS and dead tcp child >>>>>>> * Next message: [Users] Q:ABOUTE:SET UP MESSAGE FLOW >>>>>>> * Messages sorted by: [ date ] [ thread ] [ subject ] [ >>>>>>> author ] >>>>>>> >>>>>>> Hi Frank, >>>>>>> >>>>>>> can you pack the cores + binaries (lib + exe) + sources+logs(in >>>>>>> debug >>>>>>> mode) and put it somewhere for download? >>>>>>> I will try to take a look. >>>>>>> >>>>>>> regards, >>>>>>> Bogdan >>>>>>> >>>>>>> Frank DInnocenzo wrote: >>>>>>> >>>>>>> > Bogdan, >>>>>>> > >>>>>>> > I am using openser-1.1.0-tls_src. I actually get two core >>>>>>> files after >>>>>>> > the crash. Can I post them somehow? >>>>>>> > >>>>>>> > >>>>>>> > # openser -V >>>>>>> > version: openser 1.1.0-tls (i386/linux) >>>>>>> > flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, >>>>>>> > USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, >>>>>>> > FAST_LOCK-ADAPTIVE_WAIT >>>>>>> > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, >>>>>>> MAX_LISTEN 16, >>>>>>> > MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>>>>> > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>>>>>> > @(#) $Id: main.c,v 1.20 2006/07/04 17:25:54 bogdan_iancu Exp $ >>>>>>> > main.c compiled on 14:27:50 Oct 24 2006 with gcc 4.0.0 >>>>>>> > >>>>>>> > Thanks >>>>>>> > Frank >>>>>>> > >>>>>>> > >>>>>>> > */Bogdan-Andrei Iancu <bogdan at voice-system.ro>/* wrote: >>>>>>> > >>>>>>> > Hi Frank, >>>>>>> > >>>>>>> > I'm successfully using a SNOM 320 with TLS and no problem >>>>>>> so far..... >>>>>>> > What openser version are you using?...do you get any core >>>>>>> file >>>>>>> > after the >>>>>>> > crash?? >>>>>>> > >>>>>>> > regards, >>>>>>> > bogdan >>>>>>> > >>>>>>> > Frank DInnocenzo wrote: >>>>>>> > >>>>>>> > > Hello, >>>>>>> > > >>>>>>> > > I'm new to openSer and wondered if there was any >>>>>>> resolution to this >>>>>>> > > post??? I'm hitting this same crash with openSer (with >>>>>>> TLS enabled) >>>>>>> > > using a Snom phone (300 with latest Beta firmware - >>>>>>> > > snom300-6.5-SIP-j.bin ). I've gone >>>>>>> > > through the archives looking for help getting this setup >>>>>>> > working. I've >>>>>>> > > also tried the Snom Softphone with the same result. >>>>>>> > > >>>>>>> > > Thanks >>>>>>> > > Frank >>>>>>> > > >>>>>>> > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>> > > >>>>>>> > > Hi, >>>>>>> > > >>>>>>> > > I have tried to register to OpenSER CVS 31.03 with my >>>>>>> SNOM 360 >>>>>>> > > (5.3.6sw), disabled pretty all modules from OpenSER >>>>>>> (radius, >>>>>>> > xlog, auth, >>>>>>> > > ...) and always the server crashes. >>>>>>> > > >>>>>>> > > Is there something I have missed in configuration ? >>>>>>> > > >>>>>>> > > Thanks a lot for help, >>>>>>> > > -Mika >>>>>>> > > >>>>>>> > > The error message and debug info seems like this: >>>>>>> > > >>>>>>> > > 11(24351) tcpconn_new: new tcp connection to: XXX.XX.XX.XXX >>>>>>> > > 11(24351) tcpconn_new: on port 2171, type 3 >>>>>>> > > 11(24351) tls_tcpconn_init: Entered: Creating a whole >>>>>>> new ssl >>>>>>> > connection >>>>>>> > > 11(24351) tls_tcpconn_init: Looking up tls domain >>>>>>> > [XXX.XX.XX.XXX:5061] >>>>>>> > > 11(24351) tls_tcpconn_init: Using default tls server >>>>>>> settings >>>>>>> > > 11(24351) tls_tcpconn_init: Setting in ACCEPT mode (server) >>>>>>> > > 11(24351) tcpconn_add: hashes: 442, 1 >>>>>>> > > 11(24351) handle_new_connect: new connection: 0xb608f6a0 24 >>>>>>> > flags: 0002 >>>>>>> > > 11(24351) send2child: to tcp child 0 7(24347), 0xb608f6a0 >>>>>>> > > 7(24347) received n=4 con=0xb608f6a0, fd=19 >>>>>>> > > 7(24347) DBG: io_watch_add(0x810e580, 19, 2, >>>>>>> 0xb608f6a0), fd_no=1 >>>>>>> > > 7(24347) tls_update_fd: New fd is 19 >>>>>>> > > 11(24351) DBG: handle_tcp_child: dead tcp child 0 (pid >>>>>>> 24347, no 7) >>>>>>> > > (shutting down?) >>>>>>> > > 11(24351) DBG: io_watch_del (0x810e420, 17, -1, 0x0) >>>>>>> fd_no=16 called >>>>>>> > > 11(24351) ERROR: receive_fd: EOF on 15 >>>>>>> > > 11(24351) DBG: handle_ser_child: dead child 7, pid 24347 >>>>>>> > (shutting down?) >>>>>>> > > 11(24351) DBG: io_watch_del (0x810e420, 15, -1, 0x0) >>>>>>> fd_no=15 called >>>>>>> > > 0(24339) child process 24347 exited by a signal 11 >>>>>>> > > 0(24339) core was generated >>>>>>> > > 0(24339) INFO: terminating due to SIGCHLD >>>>>>> > > 6(24346) INFO: signal 15 received >>>>>>> > > 6(24346) Memory status (pkg): >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> Users(a)openser.org >>>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>> >>>> >>>> >>> >>> >> > >

3 4

Re: [Serusers] Dial Plans
by Steve Blair 06 Mar '07

06 Mar '07

Ben: You can try the following. I did not check it for accuracy but the basic idea should be correct. -Steve if (uri=~"^sip:[0-9]{4}@your-sip-domain") { prefix("111222"); if (does_uri_exist()) { strip (6); lookup("location"); t_relay(); break; } else { xlog("L_INFO", "\n[SER]: [%Tf] Subscriber check failed for <%fu> - declined\n"); sl_send_reply("603", "Declined - subscriber unavailable"); break; }; } else { .... break; }; You could prefix the xxxx with 111222, check if the new username is a subscriber and if so strip the 111222 then t_relay. POON Ben wrote: > I'll give that a try, thanks. I'm new to the whole Unix thing, so it's > taking me a while to understand what's going on... > > Thanks > Ben > > -----Original Message----- > From: Steve Blair [mailto:blairs@isc.upenn.edu] > Sent: Tuesday, March 06, 2007 9:05 AM > To: POON Ben > Subject: Re: [Serusers] Dial Plans > > > > POON Ben wrote: > >> Thanks for the quick reply. If I use alias, then I have to make an >> alias for each user, is that correct? I was wondering if there's a >> dynamic way of doing it so I don't have to make a new alias every >> > time. > >> >> > Yes. I can think of several ways but they would all involve checking > some database value. Just striping the "111222" off of the R-URI won't > do it because the resultant username does not exist in the location > table. > > You could prefix the xxxx with 111222, check if the new username is a > subscriber and if so strip the 111222 then t_relay. > > -Steve > >> Thanks, >> Ben >> >> -----Original Message----- >> From: Steve Blair [mailto:blairs@isc.upenn.edu] >> Sent: Tuesday, March 06, 2007 8:56 AM >> To: POON Ben >> Cc: serusers(a)lists.iptel.org >> Subject: Re: [Serusers] Dial Plans >> >> >> You could add an alias for these numbers. Then forward based on the >> result of a lookup"aliases". >> >> POON Ben wrote: >> >> >>> Hi, >>> >>> I need help with setting up a dial plan. >>> >>> Basically, I want to do this: >>> If my phone #'s are 111-222-xxxx where xxxx is the extension, I want >>> to be able to call xxxx instead of the full 10 digits when calling >>> within the local loop. >>> >>> I tried looked for dial plans with SER online but found very little >>> information. Any help is greatly appreciated. >>> >>> I've tried using prefix option without luck. This is what I had: >>> if(!uri=~"sip:111.*") >>> { >>> prefix("111222"); >>> }; >>> >>> When I try to call using just the extension, I get a busy tone. Are >>> there any special modules I need? And are there any good tutorials >>> on >>> >>> >> >> >>> creating dial plans and how to set them up? >>> >>> I hope I posted the question to the right place, if not please let me >>> > > >>> know where I should ask. >>> >>> Thank you very much, >>> Ben >>> --------------------------------------------------------------------- >>> - >>> -- >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >> >> > >

1 0

[Serusers] Dial Plans
by POON Ben 06 Mar '07

06 Mar '07

Hi, I need help with setting up a dial plan. Basically, I want to do this: If my phone #'s are 111-222-xxxx where xxxx is the extension, I want to be able to call xxxx instead of the full 10 digits when calling within the local loop. I tried looked for dial plans with SER online but found very little information. Any help is greatly appreciated. I've tried using prefix option without luck. This is what I had: if(!uri=~"sip:111.*") { prefix("111222"); }; When I try to call using just the extension, I get a busy tone. Are there any special modules I need? And are there any good tutorials on creating dial plans and how to set them up? I hope I posted the question to the right place, if not please let me know where I should ask. Thank you very much, Ben

3 3

[Users] click2dial
by Helmut Kuper 06 Mar '07

06 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, has anyone installed click2dial on wesip? I looking for some hints how to do that. Do I need an apache instance for it? Or it is just copy .sar file into wesipapps directory and call click2dial.jsp with a browser? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF7Tl94tZeNddg3dwRAvClAJ0Z8MluShVTpq9IGR1e1uSz6XG50ACgiVfq GvuPF087e2FXV78pvA5rLLI= =DmnR -----END PGP SIGNATURE-----

2 7

[Serusers] unable to register SIP users after 24 hours (1 day)
by Ranjeet Kumar 06 Mar '07

06 Mar '07

Hi, I am using ser 0.9.6-2 on ubuntu edgy (linux) OS. After one day, it is not registering any user what may be the cause of this problem. Is it(daemon) stops/sleeps after some time of inactivity? Thanks, Ranjeet The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the individual or entity to whom they are addressed and is subject to legal privilege. If you have received this e-mail in error you should notify the sender immediately by reply e-mail, delete the message from your system and notify your system manager. Please do not copy it for any purpose, or disclose its contents to any other person. The views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of the company. The recipient should check this e-mail and any attachments for the presence of viruses. The company accepts no liability for any damage caused, directly or indirectly, by any virus transmitted in this email. www.aztecsoft.com

1 0

[Users] Subscriber table for GUI authentication
by alper ozbilen 06 Mar '07

06 Mar '07

Dear All, I am preparing php interface for web user authentication. But some fields on subscriber table confuse my mind. please let me write below : http://openser.org/dokuwiki/doku.php/database:1.2.x#subscriber 1st: How can we generate phplib_id value to keep phplib_id field? 2nd: I cant decide on username field whether I should generate username automatically or allow users to choose any available name or number ? 3th: what is the difference between username field and phone field at last: What is meaning of ha1 and ha1b field? any help wolud be appreciated Thanks a lot Alper Ozbilen _________________________________________________________________ Rates near 39yr lows! $430K Loan for $1,399/mo - Paying Too Much? Calculate new payment http://www.lowermybills.com/lre/index.jsp?sourceid=lmb-9632-18226&moid=7581

3 2

[Serusers] ser 0.10.99 - unixsock_server.c:274: error: ‘last_process’ undeclared (first use in this function)
by "tzieleniewski" 06 Mar '07

06 Mar '07

Hi After a long way of trying to compile ser 0.10.99 I am still left some errors: Please help me with those: ser@tomix:~/src/ser-0.10.99$ make group_include="standard standard-dep stable" include_modules="xcap" all > ser.makeall.log make[1]: *** Nie podano obiektów i nie znaleziono makefile. Stop. make[1]: *** Nie podano obiektów i nie znaleziono makefile. Stop. make[1]: *** Nie podano obiektów i nie znaleziono makefile. Stop. unixsock_server.c: In function ‘init_unixsock_children’: unixsock_server.c:274: error: ‘last_process’ undeclared (first use in this function) unixsock_server.c:274: error: (Each undeclared identifier is reported only once unixsock_server.c:274: error: for each function it appears in.) unixsock_server.c:256: warning: unused variable ‘sockfd’ unixsock_server.c:253: warning: unused variable ‘backup’ make[1]: *** [unixsock_server.o] Błąd 1 checks.c: In function ‘generate_avps’: checks.c:89: error: ‘A_SER_ATTRS’ undeclared (first use in this function) checks.c:89: error: (Each undeclared identifier is reported only once checks.c:89: error: for each function it appears in.) checks.c: In function ‘radius_does_uri_exist’: checks.c:146: error: ‘V_CALL_CHECK’ undeclared (first use in this function) make[1]: *** [checks.o] Błąd 1 Errors in the above files correspond to the following lines: unixsock_server.c:274: error: snprintf(pt[last_process].desc, MAX_PT_DESC, "unix domain socket server @ %s", unixsock_name); checks.c:89: error: ‘A_SER_ATTRS’ undeclared (first use in this function) while ((vp = rc_avpair_get(vp, attrs[A_SER_ATTRS].v, 0))) { Bests Tomasz

3 2

[Users] Dialog module bug?
by T.R. Missner 06 Mar '07

06 Mar '07

Hello, We are seeing this line in our logs quite consistently. Using dialog module with openser 1.1.1 I looked at the code and cannot determine why this condition would be happening. Any ideas? BUG:dialog:update_dlg_timer: links are nul (not in timer)! Thanks TR

2 2

Re: [Serusers] Error Message
by Greger V. Teigre 06 Mar '07

06 Mar '07

Please copy the list. Strange, could you please post all the information about what you have done, your hardware configuration, your config, and the error trace to serdev? g-) Barton Fisher wrote: > Same results. I tried to compile after removing DSHM_MMAP from > Makefile.defs - no luck there either > That was only change I made after hello world test > > Bart > > Greger V. Teigre wrote: >> Hm, try a minimum (hello world) ser.cfg to see if you get the same >> error. PTR is not required. >> I assume you have not done any makefile changes. >> g-) >> >> Barton Fisher wrote: >>> Greger V. Teigre wrote: >>>> What is confusing? The message is pretty clear. But maybe the >>>> reason is unknown? >>>> What kind of system is this? Hardware? How much memory, etc etc? >>>> g-) >>>> >>>> >>>> Barton Fisher wrote: >>>>> A bit confused by startup message: >>>>> >>>>> 0(16102) ERROR: shm_mem_init: could not attach shared memory >>>>> segment: No space left on device >>>>> 0(16102) could not initialize shared memory pool, exiting... Too >>>>> much shared memory demanded: 33554432 >>>>> >>>>> I found prior message to upgrade Kernel to 2.4 - currently I'm on >>>>> 2.6.8-022stab076.1-enterprise >>>>> At another place it said to recompile ser with SYSV shm instead of >>>>> mmap (remove -DSHM_MMAP from Makefile.defs) <= This I'm not sure >>>>> exactly how / what to change in Makefile.def. >>>>> And yet another regarding no PTR <= I've contracted ISP regarding >>>>> this, but was hoping to bypass - I was hoping there was some >>>>> temporary fix for this until they act? >>>>> >>>>> I'm running SER 0.9.6 >>>>> >>>>> Thanks for any assistance. >>>>> >>>>> Bart >>>>> >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> Serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers >>> I have never seen PC, but here's what I know about it: >>> Processors = 4 >>> Model = Intel(R) XEON(TM) CPU 1.80GHz >>> Speed = 1.8GHz >>> Kernel Version = 2.6.8-022stab076.1-enterprise (SMP) >>> Distro Name = CentOS release 4.4 (Final) >>> Physical Memory = 3.89G >>> Me thinks issue is there is no PTR record as I see something like >>> this before it errors out: >>> >>> ... lots of junk ahead of this.... >>> 0(13373) if ( 0(13373) NOT( 0(13373) OR( 0(13373) uri 0(13373) =~ >>> 0(13373) "1 >>> 83" 0(13373) , 0(13373) uri 0(13373) =~ 0(13373) "200" 0(13373) ) >>> 0(13373) ) >>> 0(13373) ) { 0(13373) drop( 0(13373) type<0> 0(13373) ); 0(13373) } >>> else { 0(13 >>> 373) }; 0(13373) external_module_call( 0(13373) f_ptr<0x40573860> >>> 0(13373) , " >>> FA" 0(13373) ); 0(13373) external_module_call( 0(13373) >>> f_ptr<0x40573860> 0(13 >>> 373) , "FA" 0(13373) ); 0(13373) >>> >>> If it is PTR record checking, can I disable this checking? >>> Bart >>> >>> >>> >> >> >> >> __________ NOD32 2096 (20070305) Information __________ >> >> This message was checked by NOD32 antivirus system. >> http://www.eset.com >> >> >> > > > >

1 0

[Users] Param in SIP header
by Michel Bensoussan 06 Mar '07

06 Mar '07

2 2

[Serusers] re: npid=yes
by Alex Fler 05 Mar '07

05 Mar '07

Hello guys, Does anybody knows how to remove ";npdi=yes" from INVITE. INVITE sip:XXXXXXXXXXXXX;npdi=yes@server ip:5060;dtg=SIP;user=phone I've hear that subst_uri ,can be used . Can someone give me an example Thanks Alex Fler _____ I've stopped 5,370 spam and fraud messages. You can too! Free trial of spam and fraud protection at <http://www.cloudmark.com/sigs?rc=mrcr6> www.cloudmark.com <http://www.cloudmark.com/sigs?rc=mrcr6> Cloudmark Desktop - Join the fight against spam!

1 0

Re: [Serusers] Error Message
by Greger V. Teigre 05 Mar '07

05 Mar '07

Hm, try a minimum (hello world) ser.cfg to see if you get the same error. PTR is not required. I assume you have not done any makefile changes. g-) Barton Fisher wrote: > Greger V. Teigre wrote: >> What is confusing? The message is pretty clear. But maybe the reason >> is unknown? >> What kind of system is this? Hardware? How much memory, etc etc? >> g-) >> >> >> Barton Fisher wrote: >>> A bit confused by startup message: >>> >>> 0(16102) ERROR: shm_mem_init: could not attach shared memory >>> segment: No space left on device >>> 0(16102) could not initialize shared memory pool, exiting... Too >>> much shared memory demanded: 33554432 >>> >>> I found prior message to upgrade Kernel to 2.4 - currently I'm on >>> 2.6.8-022stab076.1-enterprise >>> At another place it said to recompile ser with SYSV shm instead of >>> mmap (remove -DSHM_MMAP from Makefile.defs) <= This I'm not sure >>> exactly how / what to change in Makefile.def. >>> And yet another regarding no PTR <= I've contracted ISP regarding >>> this, but was hoping to bypass - I was hoping there was some >>> temporary fix for this until they act? >>> >>> I'm running SER 0.9.6 >>> >>> Thanks for any assistance. >>> >>> Bart >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers > I have never seen PC, but here's what I know about it: > Processors = 4 > Model = Intel(R) XEON(TM) CPU 1.80GHz > Speed = 1.8GHz > Kernel Version = 2.6.8-022stab076.1-enterprise (SMP) > Distro Name = CentOS release 4.4 (Final) > Physical Memory = 3.89G > Me thinks issue is there is no PTR record as I see something like this > before it errors out: > > ... lots of junk ahead of this.... > 0(13373) if ( 0(13373) NOT( 0(13373) OR( 0(13373) uri 0(13373) =~ > 0(13373) "1 > 83" 0(13373) , 0(13373) uri 0(13373) =~ 0(13373) "200" 0(13373) ) > 0(13373) ) > 0(13373) ) { 0(13373) drop( 0(13373) type<0> 0(13373) ); 0(13373) } > else { 0(13 > 373) }; 0(13373) external_module_call( 0(13373) f_ptr<0x40573860> > 0(13373) , " > FA" 0(13373) ); 0(13373) external_module_call( 0(13373) > f_ptr<0x40573860> 0(13 > 373) , "FA" 0(13373) ); 0(13373) > > If it is PTR record checking, can I disable this checking? > Bart > > >

1 0

[Serusers] 4 Rings and Fast Busy
by Ing. Marco Moran 05 Mar '07

05 Mar '07

Hi, I am having an issue when I call or someone calls among the registered users, SER only gives 4 rings to answer before it gives a fast-busy. Any suggestions? Thanks, Marco -----Mensaje original----- De: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] En nombre de serusers-request(a)lists.iptel.org Enviado el: Monday, March 05, 2007 3:40 AM Para: serusers(a)lists.iptel.org Asunto: Serusers Digest, Vol 47, Issue 5 Send Serusers mailing list submissions to serusers(a)lists.iptel.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.iptel.org/mailman/listinfo/serusers or, via email, send a message with subject or body 'help' to serusers-request(a)lists.iptel.org You can reach the person managing the list at serusers-owner(a)lists.iptel.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Serusers digest..." Today's Topics: 1. Re: SIP SER (Greger V. Teigre) 2. Re: ask for ser performance parameters (Greger V. Teigre) 3. Re: test run (Greger V. Teigre) 4. Re: PATH Module in SER (0.9.6 or 0.10.x) (Greger V. Teigre) 5. Re: Problems creating DB in SER Ottendorf (Greger V. Teigre) 6. Re: speeddial - CANCEL MESSAGE ERROR (Greger V. Teigre) 7. Re: Error compiling eval module (Greger V. Teigre) 8. Re: Error Message (Greger V. Teigre) 9. RE: SIP SER (Goh, Tee-Yong) ---------------------------------------------------------------------- Message: 1 Date: Mon, 05 Mar 2007 08:02:52 +0100 From: "Greger V. Teigre" <greger(a)teigre.com> Subject: Re: [Serusers] SIP SER To: "Goh, Tee-Yong" <gohty(a)m1.com.sg> Cc: serusers(a)lists.iptel.org Message-ID: <45EBC09C.7050800(a)teigre.com> Content-Type: text/plain; charset="iso-8859-1" You can do that using exec command from ser.cfg when you receive a REGISTER. Use sipp or the internal SER stack to create new messages (see faqs on iptel.org) Or http://iptel.org/sems if you want to do it in a better way. g-) Goh, Tee-Yong wrote: > Hi Karim, > > Thank you for your advice. Noted your advice. > > Yes. My objective here is once the SER receives a REGISTER message > from one of the client, it will auto generate "INVITE" messages to the > clients. > > I am using the ser-0.9.6. I have looked through the coding as such > > /ser-0.9.6/modules/nathelper/nathelper.c > /ser-0.9.6/modules/nathelper/nathelper.cfg > /ser-0.9.6/modules/nathelper/nathelper_rrtpp.cfg > /ser-0.9.6/modules/registrar/save.c > /ser-0.9.6/modules/registrar/reply.c > /ser-0.9.6/modules/registrar/sip_msg.c > > Anyone knows where can i find the portion where once the SER receives > the REGISTER message, it responds with 200 OK to the client? Can i > find them inside nathelper or registrar or paser? > > Apologies if any inconvenience caused. > > Thank you. > > Regards, > Tee Yong > > > ------------------------------------------------------------------------ > *From:* karim basraoui [mailto:basraouik@gmail.com] > *Sent:* Sat 3/3/2007 8:36 PM > *To:* Goh, Tee-Yong > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] SIP SER > > Well, first of all a SER server does not generate messages it only > interprets the messages it receives. so it can not generate INVITE > messages. so if you want to call several clients from one client, this > one has to initiate the call by sending an INVITE to SER. > > When receiving the INVITE message, SER will forward it to the callees. > i did not try this yet, but i think that you can do this with the > method : "forward" as follows: > > if(method == INVITE) { > forward("callee1_IP_address","callee1_port"); > forward("callee2_IP_address","callee2_port"); > forward("callee3_IP_address","callee3_port"); > #and so on ... > } > > try this for example... and let me know if it works or not. > > ps: (personnally i think it won't work :p) > > 2007/3/3, Goh, Tee-Yong < gohty(a)m1.com.sg <mailto:gohty@m1.com.sg>>: > > Hi Karim, > > Yes you are right. > Please advice. > > Thank you. > > Regards, > Howard > > -----Original Message----- > *From:* karim basraoui [mailto:basraouik@gmail.com > <mailto:basraouik@gmail.com>] > *Sent:* Saturday, March 03, 2007 8:01 PM > *To:* Goh, Tee-Yong > *Cc:* serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > *Subject:* Re: [Serusers] SIP SER > > So if i understand, when a clint registers, SER will > automatically send invites to the other clients and make > connections with them? > > 2007/3/3, Goh, Tee-Yong < gohty(a)m1.com.sg > <mailto:gohty@m1.com.sg>>: > > Dear Karim, > > Thank you for your reply. > > Yes you are right. I have having difficulty searching for > this for past 2 months. > > Yes. Here is what i want. Once i receive the REGISTER > messages, my SER will send out "INVITE" message to my > other clients. This is what i am intended to do. However, > i could not search for the right coding inside the ser-0.9.6. > > Please advice and apologies if any inconvenience caused > for you. > > Thanks. > > Regards, > Howard > > -----Original Message----- > *From:* karim basraoui [mailto:basraouik@gmail.com > <mailto:basraouik@gmail.com>] > *Sent:* Saturday, March 03, 2007 7:18 PM > *To:* Goh, Tee-Yong > *Subject:* Re: [Serusers] SIP SER > > Hi, > > Do you want to make changes on SER behaviour when it > receives REGISTER messages. If it's the case you can > make a condition in the "ser.cfg" file, in the bloc > "route{ ... }". for example: > route { > if (method == "REGISTER") > { #what you want to do}; > } > > > 2007/3/3, Goh, Tee-Yong <gohty(a)m1.com.sg > <mailto:gohty@m1.com.sg>>: > > Hi, > > May i know from the SER-0.9.6, where can i look > for and make changes for the SER when receiving > "REGISTER" message and responsing " 200 OK" back > to the corresponding node or clients? > > Would appreciate your assistance and advice as > soon as possible as i have difficultly locating > the coding. Please advice where i can find the > script to edit the above mentioned messages. > > Apologies for the inconvenience caused. > > Thank you. > > Regards, > Howard > > ------------------------------------------------------------------------ > This email (including any attachment) is subject > to the following disclaimer: > http://m1.com.sg/M1/misc/disclaimer > <http://m1.com.sg/M1/misc/disclaimer> > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net > <http://www.basraouik.afrikart.net> > > ------------------------------------------------------------------------ > This email (including any attachment) is subject to the > following disclaimer: http://m1.com.sg/M1/misc/disclaimer > <http://m1.com.sg/M1/misc/disclaimer> > ------------------------------------------------------------------------ > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net <http://www.basraouik.afrikart.net> > > ------------------------------------------------------------------------ > This email (including any attachment) is subject to the following > disclaimer: http://m1.com.sg/M1/misc/disclaimer > ------------------------------------------------------------------------ > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net <http://www.basraouik.afrikart.net> > ------------------------------------------------------------------------ > This email (including any attachment) is subject to the following > disclaimer: http://m1.com.sg/M1/misc/disclaimer > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Users] avp_check and AVP's with index's?
by Brandon Armstead 05 Mar '07

05 Mar '07

Hello, Can I simply do something like so? if(avp_check("$avp(i:1[1])", "eq/s:5/g")) { # true } else { # false } ?? I seem to get an error when I try to check the value against an AVP with index's, is there an easier way then copying the index's value into a full AVP, can't I just use the index's like normal AVP's? Thanks, all help is appreciated! -- Brandon Armstead

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Re: [Serusers] PATH Module in SER (0.9.6 or 0.10.x)
by SER LIST 05 Mar '07

05 Mar '07

How do I download the early path module for 0.9? Where did you post the patch? Thanks much in advance. >From: "Greger V. Teigre" <greger(a)teigre.com> >To: SER LIST <sergrp(a)hotmail.com> >CC: serusers(a)iptel.org >Subject: Re: [Serusers] PATH Module in SER (0.9.6 or 0.10.x) >Date: Mon, 05 Mar 2007 08:16:41 +0100 > >There's an early path module in experimental for 0.9. Also, a patch was >posted to serdev by Andreas Granig. For 0.10, AFAIK there is no path >support. I assume the patch can be ported without too much work. >I have reposted the patch and asked about status on serdev. >g-) > >SER LIST wrote: >>Is there a paths module for SER? I want to implement a Load balancer using >>Dispatcher module and I will need a path module for handling >>registrations. >>Thanks in advance >> >>_________________________________________________________________ >>http://local.live.com/default.aspx?v=2&cp=43.658648~-79.383962&style=r&lvl=… >> >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> _________________________________________________________________ Free Alerts : Be smart - let your information find you ! http://alerts.live.com/Alerts/Default.aspx

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[Users] WeSIP part 2
by Helmut Kuper 05 Mar '07

05 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I was too blind to see that binaries resp. startup scripts ... :( When I'm starting WeSIP I get this output: - ------------------------------------------------- hk@vm-en-test0:~/wesip/wesip/bin$ ./startup.sh Starting WESIP Application Server VozTelecom... hk@vm-en-test0:~/wesip/wesip/bin$ HttpConnector [main] - The HttpConnector [all:8080] has been created SipConnector [main] - The SipConnector [localhost] can not be launched. Revise the configuration. SipConnector [main] - javax.sip.PeerUnavailableException: The Peer SIP Stack: com.voztele.javax.sip.SipStackImpl could not be instantiated. Ensure the Path Name has been set. javax.sip.PeerUnavailableException: The Peer SIP Stack: com.voztele.javax.sip.SipStackImpl could not be instantiated. Ensure the Path Name has been set. at javax.sip.SipFactory.createStack(SipFactory.java:308) at javax.sip.SipFactory.createSipStack(SipFactory.java:147) at com.voztele.sipservlet.connector.SipConnector.start(SipConnector.java:238) at org.apache.catalina.core.StandardService.start(StandardService.java:454) at org.apache.catalina.core.StandardServer.start(StandardServer.java:2144) at org.apache.catalina.startup.Catalina.start(Catalina.java:471) at org.apache.catalina.startup.Catalina.execute(Catalina.java:358) at org.apache.catalina.startup.Catalina.process(Catalina.java:129) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(Unknown Source) at sun.reflect.DelegatingMethodAccessorImpl.invoke(Unknown Source) at java.lang.reflect.Method.invoke(Unknown Source) at org.apache.catalina.startup.Bootstrap.main(Bootstrap.java:156) Caused by: java.lang.reflect.InvocationTargetException at sun.reflect.NativeConstructorAccessorImpl.newInstance0(Native Method) at sun.reflect.NativeConstructorAccessorImpl.newInstance(Unknown Source) at sun.reflect.DelegatingConstructorAccessorImpl.newInstance(Unknown Source) at java.lang.reflect.Constructor.newInstance(Unknown Source) at javax.sip.SipFactory.createStack(SipFactory.java:302) ... 12 more Caused by: javax.sip.PeerUnavailableException: stack name is missing at com.voztele.javax.sip.SipStackImpl.<init>(SipStackImpl.java:292) ... 17 more after a few seconds the comes a second part: ConvergedHostDeployer [main] - Installing web application at context path from URL file:/home/hk/wesip/wesip/wesipapps/ROOT processJars: cannot find /WEB-INF/lib ConvergedContext [main] - ConvergedContext [Welcome] has been started StandardService [main] - > Default Wesip Application ==> http://localhost:8080/ > Manager Wesip Application ==> http://localhost:8080/manager/html? StandardService [main] - > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml - ----------------------------------------------------------------------- my server.cml config is <Server port="5005" shutdown="SHUTDOWN" debug="0"> <Service name="WeSIP-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="8080" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="10" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="10" addresses="localhost:5081" pathName="com.voztele" protocol="udp" > <ExtraProperties> <Property key="com.voztele.javax.sip.SER_ADDRESS" value="212.6.123.118" /> <Property key="com.voztele.javax.sip.SER_PORT" value="5081" /> <Property key="com.voztele.STATISTICS" value="false" /> <Property key="com.voztele.javax.sip.STACK_NAME" value="wesip_app_server_1" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="10" /> <Property key="com.voztele.javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="com.voztele.javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="com.voztele.javax.sip.TRACE_LEVEL" value="0" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="212.6.123.118" debug="1" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.0"/> <Mapper className="com.voztele.sipservlet.core.EngineSipMapper" protocol="SIP/2.0"/> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Realm className="org.apache.catalina.realm.MemoryRealm" /> <Host name="localhost" nameSip="localhost_5081" debug="0" appBase="wesipapps" className="com.voztele.sipservlet.core.ConvergedHost"> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Mapper className="com.voztele.sipservlet.core.SipHostMapper" protocol="SIP/2.0"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.0"/>  </Host> </Engine> </Service> </Server> - --------------------------------------------------------------------------- Have no idea how to solve this java execption. The missed "/WEB-INF/lib" isn't in /home/hk/wesip/wesip/wesipapps/ROOT but in /home/hk/wesip/wesip/server/webapps/manager ... Can I copy that directory ? Helmut - -- Mit freundlichen Grüßen Helmut Kuper Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper@ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Straße 310 26133 Oldenburg Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Geschäftsführung: Hans-Joachim Iken, Dirk Thole, Jürgen Wehlend Homepage: http://www.ewetel.de ___________________________________ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF6+qR4tZeNddg3dwRAp1UAJsFkND1KaTByXA45mb8I2QkRgPjeACgv//R Y5kPDUongenLk3nZxmA2fRI= =NwKb -----END PGP SIGNATURE-----

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RE: [Serusers] SIP SER
by Goh, Tee-Yong 05 Mar '07

05 Mar '07

Dear Greger, Thank you for your advice. May i seek your advice where can i find the exec command from ser.cfg? Is there any reference? Thanks. Regards, Howard -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, March 05, 2007 3:03 PM To: Goh, Tee-Yong Cc: karim basraoui; serusers(a)lists.iptel.org Subject: Re: [Serusers] SIP SER You can do that using exec command from ser.cfg when you receive a REGISTER. Use sipp or the internal SER stack to create new messages (see faqs on iptel.org) Or http://iptel.org/sems if you want to do it in a better way. g-) Goh, Tee-Yong wrote: Hi Karim, Thank you for your advice. Noted your advice. Yes. My objective here is once the SER receives a REGISTER message from one of the client, it will auto generate "INVITE" messages to the clients. I am using the ser-0.9.6. I have looked through the coding as such /ser-0.9.6/modules/nathelper/nathelper.c /ser-0.9.6/modules/nathelper/nathelper.cfg /ser-0.9.6/modules/nathelper/nathelper_rrtpp.cfg /ser-0.9.6/modules/registrar/save.c /ser-0.9.6/modules/registrar/reply.c /ser-0.9.6/modules/registrar/sip_msg.c Anyone knows where can i find the portion where once the SER receives the REGISTER message, it responds with 200 OK to the client? Can i find them inside nathelper or registrar or paser? Apologies if any inconvenience caused. Thank you. Regards, Tee Yong _____ From: karim basraoui [ mailto:basraouik@gmail.com] Sent: Sat 3/3/2007 8:36 PM To: Goh, Tee-Yong Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SIP SER Well, first of all a SER server does not generate messages it only interprets the messages it receives. so it can not generate INVITE messages. so if you want to call several clients from one client, this one has to initiate the call by sending an INVITE to SER. When receiving the INVITE message, SER will forward it to the callees. i did not try this yet, but i think that you can do this with the method : "forward" as follows: if(method == INVITE) { forward("callee1_IP_address","callee1_port"); forward("callee2_IP_address","callee2_port"); forward("callee3_IP_address","callee3_port"); #and so on ... } try this for example... and let me know if it works or not. ps: (personnally i think it won't work :p) 2007/3/3, Goh, Tee-Yong < gohty(a)m1.com.sg>: Hi Karim, Yes you are right. Please advice. Thank you. Regards, Howard -----Original Message----- From: karim basraoui [mailto: basraouik(a)gmail.com] Sent: Saturday, March 03, 2007 8:01 PM To: Goh, Tee-Yong Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SIP SER So if i understand, when a clint registers, SER will automatically send invites to the other clients and make connections with them? 2007/3/3, Goh, Tee-Yong < <mailto:gohty@m1.com.sg> gohty(a)m1.com.sg>: Dear Karim, Thank you for your reply. Yes you are right. I have having difficulty searching for this for past 2 months. Yes. Here is what i want. Once i receive the REGISTER messages, my SER will send out "INVITE" message to my other clients. This is what i am intended to do. However, i could not search for the right coding inside the ser-0.9.6. Please advice and apologies if any inconvenience caused for you. Thanks. Regards, Howard -----Original Message----- From: karim basraoui [mailto: basraouik(a)gmail.com] Sent: Saturday, March 03, 2007 7:18 PM To: Goh, Tee-Yong Subject: Re: [Serusers] SIP SER Hi, Do you want to make changes on SER behaviour when it receives REGISTER messages. If it's the case you can make a condition in the "ser.cfg" file, in the bloc "route{ ... }". for example: route { if (method == "REGISTER") { #what you want to do}; } 2007/3/3, Goh, Tee-Yong < gohty(a)m1.com.sg <mailto:gohty@m1.com.sg> >: Hi, May i know from the SER-0.9.6, where can i look for and make changes for the SER when receiving "REGISTER" message and responsing " 200 OK" back to the corresponding node or clients? Would appreciate your assistance and advice as soon as possible as i have difficultly locating the coding. Please advice where i can find the script to edit the above mentioned messages. Apologies for the inconvenience caused. Thank you. Regards, Howard _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer <http://m1.com.sg/M1/misc/disclaimer> _____ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -- BASRAOUI Karim www.basraouik.afrikart.net _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer <http://m1.com.sg/M1/misc/disclaimer> _____ -- BASRAOUI Karim www.basraouik.afrikart.net _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer _____ -- BASRAOUI Karim www.basraouik.afrikart.net _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer _____ _____ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer

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[Serusers] Error Message
by Barton Fisher 05 Mar '07

05 Mar '07

A bit confused by startup message: 0(16102) ERROR: shm_mem_init: could not attach shared memory segment: No space left on device 0(16102) could not initialize shared memory pool, exiting... Too much shared memory demanded: 33554432 I found prior message to upgrade Kernel to 2.4 - currently I'm on 2.6.8-022stab076.1-enterprise At another place it said to recompile ser with SYSV shm instead of mmap (remove -DSHM_MMAP from Makefile.defs) <= This I'm not sure exactly how / what to change in Makefile.def. And yet another regarding no PTR <= I've contracted ISP regarding this, but was hoping to bypass - I was hoping there was some temporary fix for this until they act? I'm running SER 0.9.6 Thanks for any assistance. Bart

2 1

[Serusers] Error compiling eval module
by Andrey Kuprianov 05 Mar '07

05 Mar '07

Guys, I've just updated sip_router from CVS. Cant compile eval module no more. It says, eval.c:33:23: error: uuid/uuid.h: No such file or directory eval.c: In function 'sel_gen_uuid': eval.c:1975: error: 'uuid_t' undeclared (first use in this function) eval.c:1975: error: (Each undeclared identifier is reported only once eval.c:1975: error: for each function it appears in.) eval.c:1975: error: expected ';' before 'u' eval.c:1976: warning: implicit declaration of function 'uuid_generate' eval.c:1976: error: 'u' undeclared (first use in this function) eval.c:1977: warning: implicit declaration of function 'uuid_unparse' make[1]: *** [eval.o] Error 1 make[1]: Leaving directory `/home/andrey/sip_router/modules/eval' make: *** [modules] Error 1 Plz, help :)) Andrey.

3 3

[Serusers] speeddial - CANCEL MESSAGE ERROR
by tzieleniewski 05 Mar '07

05 Mar '07

Hi!! I was utilizing the speeddial functionality and I encountered the following error: When user sends a cancel message before the destination leg answers the call the cancel message is not properly routed, because sd_lookup("speed_dial") returns negative result for the processing of cancel message. If other side answers everything is ok (BYE etc). here is my Speed_dial logic: if (uri =~ "sip:[0-9]{2}@.*") { if(sd_lookup("speed_dial")) { route(FORWARD); drop; } } Bests tomasz

2 1

[Serusers] Problems creating DB in SER Ottendorf
by Andrey Kuprianov 05 Mar '07

05 Mar '07

Hi, Im running Ubuntu with MySQL 5.0.24a. I've downloaded and compiled Ottendorf from CVS successfully. The problem came, when I tried to create db with sip_router/scripts/mysql/ser_mysql.sh create. It asked for root password (i dont have any so I pressed Enter right away), but then displayed mysql options (give below) for me instead of creating DB. I suppose some argument went missing. How can I fix it? I also remember there was a manual way to do it, but I couldnt find a proper link. Can anyone help me, plz? Output: mysql Ver 14.12 Distrib 5.0.24a, for pc-linux-gnu (i486) using readline 5.1 Copyright (C) 2002 MySQL AB This software comes with ABSOLUTELY NO WARRANTY. This is free software, and you are welcome to modify and redistribute it under the GPL license Usage: mysql [OPTIONS] [database] -?, --help Display this help and exit. -I, --help Synonym for -? --auto-rehash Enable automatic rehashing. One doesn't need to use 'rehash' to get table and field completion, but startup and reconnecting may take a longer time. Disable with --disable-auto-rehash. -A, --no-auto-rehash No automatic rehashing. One has to use 'rehash' to get table and field completion. This gives a quicker start of mysql and disables rehashing on reconnect. WARNING: options deprecated; use --disable-auto-rehash instead. -B, --batch Don't use history file. Disable interactive behavior. (Enables --silent) --character-sets-dir=name Directory where character sets are. --default-character-set=name Set the default character set. -C, --compress Use compression in server/client protocol. -#, --debug[=#] This is a non-debug version. Catch this and exit -D, --database=name Database to use. --delimiter=name Delimiter to be used. -e, --execute=name Execute command and quit. (Disables --force and history file) -E, --vertical Print the output of a query (rows) vertically. -f, --force Continue even if we get an sql error. -g, --no-named-commands Named commands are disabled. Use \* form only, or use named commands only in the beginning of a line ending with a semicolon (;) Since version 10.9 the client now starts with this option ENABLED by default! Disable with '-G'. Long format commands still work from the first line. WARNING: option deprecated; use --disable-named-commands instead. -G, --named-commands Enable named commands. Named commands mean this program's internal commands; see mysql> help . When enabled, the named commands can be used from any line of the query, otherwise only from the first line, before an enter. Disable with --disable-named-commands. This option is disabled by default. -i, --ignore-spaces Ignore space after function names. --local-infile Enable/disable LOAD DATA LOCAL INFILE. -b, --no-beep Turn off beep on error. -h, --host=name Connect to host. -H, --html Produce HTML output. -X, --xml Produce XML output --line-numbers Write line numbers for errors. -L, --skip-line-numbers Don't write line number for errors. WARNING: -L is deprecated, use long version of this option instead. --no-pager Disable pager and print to stdout. See interactive help (\h) also. WARNING: option deprecated; use --disable-pager instead. --no-tee Disable outfile. See interactive help (\h) also. WARNING: option deprecated; use --disable-tee instead -n, --unbuffered Flush buffer after each query. --column-names Write column names in results. -N, --skip-column-names Don't write column names in results. WARNING: -N is deprecated, use long version of this options instead. -O, --set-variable=name Change the value of a variable. Please note that this option is deprecated; you can set variables directly with --variable-name=value. --sigint-ignore Ignore SIGINT (CTRL-C) -o, --one-database Only update the default database. This is useful for skipping updates to other database in the update log. --pager[=name] Pager to use to display results. If you don't supply an option the default pager is taken from your ENV variable PAGER. Valid pagers are less, more, cat [> filename], etc. See interactive help (\h) also. This option does not work in batch mode. -p, --password[=name] Password to use when connecting to server. If password is not given it's asked from the tty. WARNING: This is insecure as the password is visible for anyone through /proc for a short time. -P, --port=# Port number to use for connection. --prompt=name Set the mysql prompt to this value. --protocol=name The protocol of connection (tcp,socket,pipe,memory). -q, --quick Don't cache result, print it row by row. This may slow down the server if the output is suspended. Doesn't use history file. -r, --raw Write fields without conversion. Used with --batch. --reconnect Reconnect if the connection is lost. Disable with --disable-reconnect. This option is enabled by default. -s, --silent Be more silent. Print results with a tab as separator, each row on new line. -S, --socket=name Socket file to use for connection. --ssl Enable SSL for connection (automatically enabled with other flags). Disable with --skip-ssl. --ssl-ca=name CA file in PEM format (check OpenSSL docs, implies --ssl). --ssl-capath=name CA directory (check OpenSSL docs, implies --ssl). --ssl-cert=name X509 cert in PEM format (implies --ssl). --ssl-cipher=name SSL cipher to use (implies --ssl). --ssl-key=name X509 key in PEM format (implies --ssl). --ssl-verify-server-cert Verify server's "Common Name" in its cert against hostname used when connecting. This option is disabled by default. -t, --table Output in table format. -T, --debug-info Print some debug info at exit. --tee=name Append everything into outfile. See interactive help (\h) also. Does not work in batch mode. -u, --user=name User for login if not current user. -U, --safe-updates Only allow UPDATE and DELETE that uses keys. -U, --i-am-a-dummy Synonym for option --safe-updates, -U. -v, --verbose Write more. (-v -v -v gives the table output format). -V, --version Output version information and exit. -w, --wait Wait and retry if connection is down. --connect_timeout=# Number of seconds before connection timeout. --max_allowed_packet=# Max packet length to send to, or receive from server --net_buffer_length=# Buffer for TCP/IP and socket communication --select_limit=# Automatic limit for SELECT when using --safe-updates --max_join_size=# Automatic limit for rows in a join when using --safe-updates --secure-auth Refuse client connecting to server if it uses old (pre-4.1.1) protocol --show-warnings Show warnings after every statement. Default options are read from the following files in the given order: /etc/mysql/my.cnf ~/.my.cnf /etc/mysql/my.cnf The following groups are read: mysql client The following options may be given as the first argument: --print-defaults Print the program argument list and exit --no-defaults Don't read default options from any options file --defaults-file=# Only read default options from the given file # --defaults-extra-file=# Read this file after the global files are read Variables (--variable-name=value) and boolean options {FALSE|TRUE} Value (after reading options) --------------------------------- ----------------------------- auto-rehash TRUE character-sets-dir (No default value) default-character-set latin1 compress FALSE database (No default value) delimiter ; vertical FALSE force FALSE named-commands FALSE local-infile FALSE no-beep FALSE host localhost html FALSE xml FALSE line-numbers TRUE unbuffered FALSE column-names TRUE sigint-ignore FALSE port 3306 prompt mysql> quick FALSE raw FALSE reconnect FALSE socket /var/run/mysqld/mysqld.sock ssl FALSE ssl-ca (No default value) ssl-capath (No default value) ssl-cert (No default value) ssl-cipher (No default value) ssl-key (No default value) ssl-verify-server-cert FALSE table FALSE debug-info FALSE user root safe-updates FALSE i-am-a-dummy FALSE connect_timeout 0 max_allowed_packet 16777216 net_buffer_length 16384 select_limit 1000 max_join_size 1000000 secure-auth FALSE show-warnings FALSE Andrey.

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[Serusers] PATH Module in SER (0.9.6 or 0.10.x)
by SER LIST 05 Mar '07

05 Mar '07

Is there a paths module for SER? I want to implement a Load balancer using Dispatcher module and I will need a path module for handling registrations. Thanks in advance _________________________________________________________________ http://local.live.com/default.aspx?v=2&cp=43.658648~-79.383962&style=r&lvl=…

2 1

[Serusers] ask for ser performance parameters
by Su Kerry 05 Mar '07

05 Mar '07

3 3

[Users] WeSIP b2bua problem
by tele 04 Mar '07

04 Mar '07

Hi all, I've read on the WeSIP website that the support is made through openser mailing-list, so here i'am. I've problem with the SIP Servlet in b2bua mode, the problem is that the ACK is not forwarded to the other call-leg. INVITE, 183 session progress, 180 ringing are correctly forwarded but the ACK no. In the debug i can't see too much information about the request is not forwarded. Attached there is a .zip with all the log, trace and configuration. thanks for the support, :tele

1 0

Re: FW: [Users] imc
by Daniel-Constantin Mierla 03 Mar '07

03 Mar '07

Hello, because the registrar module imports now some values of usrloc parameters it must be loaded before the usrloc module. http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x#registrar use: loadmodule "usrloc.so" loadmodule "registrar.so" Cheers, Daniel On 03/03/07 15:14, Henri Keski-Sikkilä wrote: > > Daniel-Constantin Mierla wrote: > > Hello, > > > > henri keski-sikkila 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 Daniel, > > I took the latest source code from svn. I used a default routing script > "imc.cfg" that comes with the imc module. When I started openser, I got > these messages: > > 0(0) imc:mod_init:the query returned no result > 0(0) find_cmd_export_t: found <load_tm>(0) in module tm > [modules/tm/tm.so] > 0(0) pool_remove: Removing connection from the pool > 0(0) DEBUG: register_stats: imc > 0(0) DEBUG: register_mi: imc > 0(0) DEBUG: init_mod: tm > 0(0) TM - initializing... > 0(0) Call-ID initialization: '7fbb9de7' > 0(0) unixsock_register_cmd: New command (t_uac_dlg) registered > 0(0) unixsock_register_cmd: New command (t_uac_cancel) registered > 0(0) unixsock_register_cmd: New command (t_hash) registered > 0(0) unixsock_register_cmd: New command (t_reply) registered > 0(0) DEBUG: lock_initialize: lock initialization started > 0(0) DEBUG: MD5 calculated: 72dbedf8f6d9bc423ac07e79fbac2ddb > 0(0) DEBUG: MD5 calculated: ce1d8ecb1b6d4462277ac0abb13e18fc > 0(0) DEBUG: register_stats: tm > 0(0) DEBUG: register_mi: tm > 0(0) DEBUG: init_mod: xlog > 0(0) XLOG: initializing ... > 0(0) DEBUG: init_mod: registrar > 0(0) registrar - initializing > 0(0) find_cmd_export_t: found <load_sl>(0) in module sl > [modules/sl/sl.so] > 0(0) find_cmd_export_t: found <ul_bind_usrloc>(1) in module usrloc > [modules/usrloc/usrloc.so] > 0(0) ERROR:usrloc:bind_usrloc: configuration error - trying to bind to > usrlo module before being initialized > 0(0) init_mod(): Error while initializing module registrar > ERROR: error while initializing modules > 0(0) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was > not allocated -> nothing to destroy > 0(0) XLOG: destroy module ... > 0(0) DEBUG: tm_shutdown : start > 0(0) DEBUG: unlink_timer_lists : emptying DELETE list > 0(0) DEBUG: tm_shutdown : emptying hash table > 0(0) DEBUG: tm_shutdown : releasing timers > 0(0) DEBUG: tm_shutdown : removing semaphores > 0(0) DEBUG: tm_shutdown : destroying tmcb lists > 0(0) DEBUG: tm_shutdown : done > 0(0) imc: destroy module ... > 0(15800) shm_mem_destroy > 0(15800) destroying the shared memory lock > > > Best Regards, > Henri > >

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Re: [Serusers] SIP SER
by karim basraoui 03 Mar '07

03 Mar '07

So if i understand, when a clint registers, SER will automatically send invites to the other clients and make connections with them? 2007/3/3, Goh, Tee-Yong <gohty(a)m1.com.sg>: > > Dear Karim, > > Thank you for your reply. > > Yes you are right. I have having difficulty searching for this for past 2 > months. > > Yes. Here is what i want. Once i receive the REGISTER messages, my SER > will send out "INVITE" message to my other clients. This is what i am > intended to do. However, i could not search for the right coding inside the > ser-0.9.6. > > Please advice and apologies if any inconvenience caused for you. > > Thanks. > > Regards, > Howard > > -----Original Message----- > *From:* karim basraoui [mailto:basraouik@gmail.com] > *Sent:* Saturday, March 03, 2007 7:18 PM > *To:* Goh, Tee-Yong > *Subject:* Re: [Serusers] SIP SER > > Hi, > > Do you want to make changes on SER behaviour when it receives REGISTER > messages. If it's the case you can make a condition in the "ser.cfg" file, > in the bloc "route{ ... }". for example: > route { > if (method == "REGISTER") > { #what you want to do}; > } > > > 2007/3/3, Goh, Tee-Yong <gohty(a)m1.com.sg >: > > > > Hi, > > > > May i know from the SER-0.9.6, where can i look for and make changes for > > the SER when receiving "REGISTER" message and responsing " 200 OK" back to > > the corresponding node or clients? > > > > Would appreciate your assistance and advice as soon as possible as i > > have difficultly locating the coding. Please advice where i can find the > > script to edit the above mentioned messages. > > > > Apologies for the inconvenience caused. > > > > Thank you. > > > > Regards, > > Howard > > > > ------------------------------ > > This email (including any attachment) is subject to the following > > disclaimer: http://m1.com.sg/M1/misc/disclaimer > > ------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net > > ------------------------------ > This email (including any attachment) is subject to the following > disclaimer: http://m1.com.sg/M1/misc/disclaimer > ------------------------------ > -- BASRAOUI Karim www.basraouik.afrikart.net

1 0

[Serusers] SIP SER
by Goh, Tee-Yong 03 Mar '07

03 Mar '07

Hi, May i know from the SER-0.9.6, where can i look for and make changes for the SER when receiving "REGISTER" message and responsing " 200 OK" back to the corresponding node or clients? Would appreciate your assistance and advice as soon as possible as i have difficultly locating the coding. Please advice where i can find the script to edit the above mentioned messages. Apologies for the inconvenience caused. Thank you. Regards, Howard This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer

1 0

[Users] regarding SIPS/TLS
by Andrew T Gin 02 Mar '07

02 Mar '07

Hi I'm not sure whether this belongs in the developer one or not. When I proxy requests through openSER, and the contact address for the client uses a secure URI (SIPS) and the protocol stated is TLS, I get the following error: 7(6639) ERROR: uri2proxy: bad transport for sips uri: 3 7(6639) ERROR:tm:t_forward_nonack: failure to add branches This is in ut.h, line 113. If the URI is a secure one (SIPS), why must the protocol be TCP or NONE? openSER exits when the URI is secure (SIPS) AND the transport is TLS. Why is this?

2 1

[Users] Module UAC to authentication and redirect calls
by Thiago Maluf 02 Mar '07

02 Mar '07

Hi List, I need config one module UAC to do authentication and redirect my calls to other Voip Provider. Somebody already did it? If can tell me what functions you used. Thanks in advanced! Thiago Maluf. ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) e-mail: malufrj(a)gmail.com

2 1

[Users] OpenSer 1.1 crashs on TLS
by Ohad.Levy＠infineon.com 02 Mar '07

02 Mar '07

Hi, I've used the default configuration for TLS, and tried to register one snom 320 phone, openser crashes immediately. Attached is a log - any ideas? Thanks, Ohad 19(14449) tcpconn_new: new tcp connection to: 172.28.129.126 19(14449) tcpconn_new: on port 2358, type 3 19(14449) tls_tcpconn_init: Entered: Creating a whole new ssl connection 19(14449) tls_tcpconn_init: Looking up socket based TLS server domain [172.29.162.26:5061] 19(14449) tls_find_server_domain: virtual TLS server domain not found, Using default TLS server domain settings 19(14449) tls_tcpconn_init: Found socket based TLS server domain [0.0.0.0:0] 19(14449) tls_tcpconn_init: Setting in ACCEPT mode (server) 19(14449) tcpconn_add: hashes: 410, 1 19(14449) handle_new_connect: new connection: 0xb60b93f8 36 flags: 0002 19(14449) send2child: to tcp child 0 11(14441), 0xb60b93f8 11(14441) received n=4 con=0xb60b93f8, fd=23 11(14441) DBG: io_watch_add(0x80fea60, 23, 2, 0xb60b93f8), fd_no=1 11(14441) tls_update_fd: New fd is 23 19(14449) ERROR: receive_fd: EOF on 18 19(14449) DBG: handle_ser_child: dead child 11, pid 14441 (shutting down?) 19(14449) DBG: io_watch_del (0x80fe900, 18, -1, 0x0) fd_no=27 called 19(14449) DBG: handle_tcp_child: dead tcp child 0 (pid 14441, no 11) (shutting down?) 19(14449) DBG: io_watch_del (0x80fe900, 20, -1, 0x0) fd_no=26 called 0(14430) child process 14441 exited by a signal 11 0(14430) core was generated 0(14430) INFO: terminating due to SIGCHLD 1(14431) INFO: signal 15 received 1(14431) Memory status (pkg): 1(14431) fm_status (0x8130780): 1(14431) heap size= 1048576 1(14431) used= 22016, used+overhead=40752, free=1007824 1(14431) max used (+overhead)= 40752 1(14431) dumping free list: 1(14431) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) 1(14431) hash = 16 fragments no.: 1, unused: 0 bucket size: 128 - 128 (first 128) 1(14431) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 1(14431) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 2(14432) INFO: signal 15 received 2(14432) Memory status (pkg): 2(14432) fm_status (0x8130780): 2(14432) heap size= 1048576 2(14432) used= 21888, used+overhead=40576, free=1008000 2(14432) max used (+overhead)= 40608 2(14432) dumping free list: 2(14432) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 2(14432) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 2(14432) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 2(14432) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 2(14432) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 2(14432) TOTAL: 6 free fragments = 1008000 free bytes 2(14432) ----------------------------- 3(14433) INFO: signal 15 received 3(14433) Memory status (pkg): 3(14433) fm_status (0x8130780): 3(14433) heap size= 1048576 3(14433) used= 21888, used+overhead=40576, free=1008000 3(14433) max used (+overhead)= 40608 3(14433) dumping free list: 3(14433) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 3(14433) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 3(14433) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 3(14433) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 3(14433) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 3(14433) TOTAL: 6 free fragments = 1008000 free bytes 3(14433) ----------------------------- 4(14434) INFO: signal 15 received 4(14434) Memory status (pkg): 4(14434) fm_status (0x8130780): 4(14434) heap size= 1048576 4(14434) used= 21888, used+overhead=40576, free=1008000 4(14434) max used (+overhead)= 40608 4(14434) dumping free list: 4(14434) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 4(14434) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 4(14434) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 4(14434) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 4(14434) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 4(14434) TOTAL: 6 free fragments = 1008000 free bytes 4(14434) ----------------------------- 5(14435) INFO: signal 15 received 5(14435) Memory status (pkg): 5(14435) fm_status (0x8130780): 5(14435) heap size= 1048576 5(14435) used= 21888, used+overhead=40576, free=1008000 5(14435) max used (+overhead)= 40608 5(14435) dumping free list: 5(14435) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 5(14435) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 5(14435) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 5(14435) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 5(14435) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 5(14435) TOTAL: 6 free fragments = 1008000 free bytes 5(14435) ----------------------------- 6(14436) INFO: signal 15 received 6(14436) Memory status (pkg): 6(14436) fm_status (0x8130780): 6(14436) heap size= 1048576 6(14436) used= 21888, used+overhead=40576, free=1008000 6(14436) max used (+overhead)= 40608 6(14436) dumping free list: 6(14436) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 6(14436) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 6(14436) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 6(14436) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 6(14436) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 6(14436) TOTAL: 6 free fragments = 1008000 free bytes 6(14436) ----------------------------- 7(14437) INFO: signal 15 received 7(14437) Memory status (pkg): 7(14437) fm_status (0x8130780): 7(14437) heap size= 1048576 7(14437) used= 21888, used+overhead=40576, free=1008000 7(14437) max used (+overhead)= 40608 7(14437) dumping free list: 7(14437) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 7(14437) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 7(14437) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 7(14437) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 7(14437) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 7(14437) TOTAL: 6 free fragments = 1008000 free bytes 7(14437) ----------------------------- 8(14438) INFO: signal 15 received 8(14438) Memory status (pkg): 8(14438) fm_status (0x8130780): 8(14438) heap size= 1048576 8(14438) used= 21888, used+overhead=40576, free=1008000 8(14438) max used (+overhead)= 40608 8(14438) dumping free list: 8(14438) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 8(14438) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 8(14438) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 8(14438) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 8(14438) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 8(14438) TOTAL: 6 free fragments = 1008000 free bytes 8(14438) ----------------------------- 9(14439) INFO: signal 15 received 9(14439) Memory status (pkg): 9(14439) fm_status (0x8130780): 9(14439) heap size= 1048576 9(14439) used= 21888, used+overhead=40576, free=1008000 9(14439) max used (+overhead)= 40608 9(14439) dumping free list: 9(14439) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 9(14439) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 9(14439) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 9(14439) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 9(14439) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 9(14439) TOTAL: 6 free fragments = 1008000 free bytes 9(14439) ----------------------------- 10(14440) INFO: signal 15 received 10(14440) Memory status (pkg): 10(14440) fm_status (0x8130780): 10(14440) heap size= 1048576 10(14440) used= 21856, used+overhead=40536, free=1008040 10(14440) max used (+overhead)= 40608 10(14440) dumping free list: 10(14440) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 10(14440) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 10(14440) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 10(14440) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 10(14440) TOTAL: 6 free fragments = 1008040 free bytes 10(14440) ----------------------------- 12(14442) INFO: signal 15 received 12(14442) Memory status (pkg): 12(14442) fm_status (0x8130780): 12(14442) heap size= 1048576 12(14442) used= 72048, used+overhead=90744, free=957832 12(14442) max used (+overhead)= 90744 12(14442) dumping free list: 12(14442) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 12(14442) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 12(14442) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 12(14442) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 12(14442) TOTAL: 6 free fragments = 957832 free bytes 12(14442) ----------------------------- 13(14443) INFO: signal 15 received 13(14443) Memory status (pkg): 13(14443) fm_status (0x8130780): 13(14443) heap size= 1048576 13(14443) used= 72048, used+overhead=90744, free=957832 13(14443) max used (+overhead)= 90744 13(14443) dumping free list: 13(14443) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 13(14443) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 13(14443) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 13(14443) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 13(14443) TOTAL: 6 free fragments = 957832 free bytes 13(14443) ----------------------------- 14(14444) INFO: signal 15 received 14(14444) Memory status (pkg): 14(14444) fm_status (0x8130780): 14(14444) heap size= 1048576 14(14444) used= 72048, used+overhead=90744, free=957832 14(14444) max used (+overhead)= 90744 14(14444) dumping free list: 14(14444) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 14(14444) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 14(14444) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 14(14444) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 14(14444) TOTAL: 6 free fragments = 957832 free bytes 14(14444) ----------------------------- 15(14445) INFO: signal 15 received 15(14445) Memory status (pkg): 15(14445) fm_status (0x8130780): 15(14445) heap size= 1048576 15(14445) used= 72048, used+overhead=90744, free=957832 15(14445) max used (+overhead)= 90744 15(14445) dumping free list: 15(14445) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 15(14445) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 15(14445) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 15(14445) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 15(14445) TOTAL: 6 free fragments = 957832 free bytes 15(14445) ----------------------------- 16(14446) INFO: signal 15 received 16(14446) Memory status (pkg): 16(14446) fm_status (0x8130780): 16(14446) heap size= 1048576 16(14446) used= 72048, used+overhead=90744, free=957832 16(14446) max used (+overhead)= 90744 16(14446) dumping free list: 16(14446) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 16(14446) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 16(14446) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 16(14446) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 16(14446) TOTAL: 6 free fragments = 957832 free bytes 16(14446) ----------------------------- 17(14447) INFO: signal 15 received 17(14447) Memory status (pkg): 17(14447) fm_status (0x8130780): 17(14447) heap size= 1048576 17(14447) used= 72048, used+overhead=90744, free=957832 17(14447) max used (+overhead)= 90744 17(14447) dumping free list: 17(14447) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 17(14447) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 17(14447) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 17(14447) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 17(14447) TOTAL: 6 free fragments = 957832 free bytes 17(14447) ----------------------------- 18(14448) INFO: signal 15 received 18(14448) Memory status (pkg): 18(14448) fm_status (0x8130780): 18(14448) heap size= 1048576 18(14448) used= 72048, used+overhead=90744, free=957832 18(14448) max used (+overhead)= 90744 18(14448) dumping free list: 18(14448) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 18(14448) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 18(14448) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 18(14448) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 18(14448) TOTAL: 6 free fragments = 957832 free bytes 18(14448) ----------------------------- 19(14449) INFO: signal 15 received 19(14449) Memory status (pkg): 19(14449) fm_status (0x8130780): 19(14449) heap size= 1048576 19(14449) used= 72048, used+overhead=90744, free=957832 19(14449) max used (+overhead)= 90744 19(14449) dumping free list: 19(14449) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 19(14449) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 19(14449) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 19(14449) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 19(14449) TOTAL: 6 free fragments = 957832 free bytes 19(14449) ----------------------------- 1(14431) TOTAL: 4 free fragments = 1007824 free bytes 1(14431) ----------------------------- 0(14430) XLOG: destroy module ... 0(14430) pool_remove: Connection still kept in the pool 0(14430) pool_remove: Connection still kept in the pool 0(14430) pool_remove: Removing connection from the pool 0(14430) DEBUG: tm_shutdown : start 0(14430) DEBUG: unlink_timer_lists : emptying DELETE list 0(14430) DEBUG: tm_shutdown : emptying hash table 0(14430) DEBUG: tm_shutdown : releasing timers 0(14430) DEBUG: tm_shutdown : removing semaphores 0(14430) DEBUG: tm_shutdown : destroying tmcb lists 0(14430) DEBUG: tm_shutdown : done 0(14430) tls_close: Closing SSL connection 0(14430) tls_update_fd: New fd is 36 0(14430) tls_shutdown: Shutdown successful 0(14430) tls_tcpconn_clean: Entered 1(14431) dumping free list:

2 1

[Users] WeSIP
by Helmut Kuper 02 Mar '07

02 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm interessting in using WeSIP as a ALG. I got running openser as a SIP frontend for WeSIP by using SEAS module. But I have no idea what do to to get WeSIP running. Do I need here apache to run those xml files? Or is there any WeSIP binaries to strat WeSIP Server ?? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD4DBQFF6Bc84tZeNddg3dwRAmchAKCtliDoToB1SeWeNEJtQ/AKf4viJgCXVZPv D6k7Vzrru5NaXEXLL70IBQ== =kpT1 -----END PGP SIGNATURE-----

2 1

[Users] WeSIP API
by raviprakash sunkara 02 Mar '07

02 Mar '07

Hello Users Good Morning, Is Any Using the WeSIP to OpenSER for SEAS, Please can anyone Give the WeSIP API documentation , So that I view the Classes and Methods in WeSIP, Does WeSIP supports the b2bua for the OpenSER, So that I can Generate it, I'll use the WeSIP only for INVITE request, For Post Paid Billing Generating Main Question is WeSIP support the NAT, -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

3 2

[Users] Some precisions about reply_route handling
by Jean-Franois SMIGIELSKI 01 Mar '07

01 Mar '07

Hello list, My openser (1.1.0) does not forward provisionnal replies. Currently, the "180 Ringing" I receive are absorbed, and that is bad. But I don't use any configuration directive to explicitely and statefully forward provisionnal answers. Please, which can I use? The reply_route block that is used is set with a call to t_on_reply . Thanks a lot. JF Smigielski.

2 1

[Users] Loose routing and branches
by Marius Patea 01 Mar '07

01 Mar '07

Hello I have an OpenSER which receives an initial INVITE request with a Route HF. It processes Route HF (by applying loose routing mechanism). Then it appends a new branch and relay the request. My problem is that the CANCEL request, which is generated by the TM module when a 200 OK response from the other leg was acknowledged, contains the original Route HF from the INVITE request, and not the processed Route HF. I think that the correct behavior is that the Route HF from the CANCEL request should not contains this proxy as topmost route. Thanks for any hints! Regards, MArius

1 0

[Serusers] SER as Presence Server
by gauri 01 Mar '07

01 Mar '07

Hi, We need Presence Server for testing purpose. Which version of SER has support for Presence Server and is it stable. Any pointers in this direction will be helpful. Thanks, Gauri 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

2 1

[Serusers] how to log unauthorized attemp with clear password
by Fabio Macchi 01 Mar '07

01 Mar '07

Hi all, I'm using ser-0.9.7 with mysql authentication. This is the INVITE handling section of my ser.cfg # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); } if (!allow_trusted()) { if (!proxy_authorize("sip.keeptelecom.com","subscriber")) { proxy_challenge("sip.keeptelecom.com","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; # ------ Logging calls after succesfully auth ----------- if (isflagset(6)) { xlog("L_INFO", "<%is> CH <%ct> R-URI <%ru> autenticato: NAT detected\n"); } else { xlog("L_INFO", "<%is> CH <%ct> R-URI <%ru> autenticato: no NAT detected\n"); } As you cas see, I've added just after authentication section some lines to log every succesfully authenticated attemp to make a call, with NAT detection info. I'd like, in the same way, add a single line to log every wrong user or wrong password attempt. My question is: which is the correct place to set this in the configuration file ? My first idea was to place xlog instruction just inside the if(!proxy_authorize){ . , but this produces multiple line in log, because in a call first invite is sent without proxy_authorization section. Is there a way, with xlog, to show the client sent password in clear for an easy troubleshoot ? Many thanks for any suggestion. Fabio

2 1

[Users] connection pooling with unixodbc
by Papadopoulos Georgios 01 Mar '07

01 Mar '07

hello all I am loading the following 10 modules that use an oracle database through unixodbc. modparam("usrloc|acc", "db_url", "unixodbc://user:pass@/OracleODBC-10g") modparam("auth_db|alias_db|uri_db|group|permissions|lcr|domain", "db_url", "unixodbc://user:pass@/OracleODBC-10g") modparam("avpops", "avp_url", "unixodbc://user:pass@/OracleODBC-10g") In my config I have fork=yes and tcp is enabled With 5 listen addresses and children=1 : 10 connections in the database. children=2 : 16 connections children=3 : 22 connections With 4 listen address and children=1 : 9 connections in the database. children=2 : 14 connections children=3 : 19 connections Could anyone help me understand the relation between listen addresses, number of children and number of modules that use the db? Also these numbers are the same whether I use MySQL both through the mysql module and the unixodbc module. Looking into the code, it seems like there is connection pooling into the unixodbc module. However I get all these connections. What am I doing wrong? thank you for any help 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.

2 1

[Serusers] voice transmitting
by zahra bahar 01 Mar '07

01 Mar '07

Hi all. I have 2 xlite phone with ser server. when establshing call only for example user3 can hear user4 voice and user4 can hear the voice of itself too!and user3 should be on speakerphone mode. what's wrong?when we should be on speaker phone? --------------------------------- The fish are biting. Get more visitors on your site using Yahoo! Search Marketing.

1 0

Re: [Users] TLS and dead tcp child
by Paulo Angonese 01 Mar '07

01 Mar '07

The core, binaries and the full log are in ftp://ftp.procergs.com.br/pub/procergs/openser This is the backtrace: .... #0 0x00c63b87 in strchr () from /lib/libc.so.6 #1 0x0033dc53 in krb5_kt_resolve () from /usr/lib/libkrb5.so.3 #2 0x003cf5a0 in kssl_keytab_is_available () from /lib/libssl.so.5 #3 0x003bbc4c in ssl3_choose_cipher () from /lib/libssl.so.5 #4 0x003b73c0 in ssl3_accept () from /lib/libssl.so.5 #5 0x003c4eba in SSL_accept () from /lib/libssl.so.5 #6 0x080e1dc9 in tls_accept (c=0xb616b490) at tls/tls_server.c:236 #7 0x080e35cc in tls_fix_read_conn (c=0xb616b490) at tls/tls_server.c:872 #8 0x0809c428 in tcp_read_req (con=0xb616b490, bytes_read=0xbfad8448) at tcp_read.c:411 #9 0x0809c7a7 in handle_io (fm=0x814e1b0, idx=-1) at tcp_read.c:772 #10 0x0809dbf3 in io_wait_loop_epoll (h=0x8113020, t=Variable "t" is not available. ) at io_wait.h:722 #11 0x0809df67 in tcp_receive_loop (unix_sock=17) at tcp_read.c:893 #12 0x08093969 in tcp_init_children (chd_rank=0x810dcd4) at tcp_main.c:1745 #13 0x0806ce1e in main_loop () at main.c:1009 #14 0x0806e8cb in main (argc=1, argv=0xbfad86f4) at main.c:1477 My version of krb5-libs is krb5-libs-1.4.1-5 Please, any help? Thanks Paulo Angonese >>> >>> Paulo Angonese wrote: >>>> Ok. >>>> >>>> The core (two files was generated) and the full log (debug=9) are in >>>> ftp://ftp.procergs.com.br/pub/procergs/openser >>>> >>>> >>>> >>>> Bogdan-Andrei Iancu escreveu: >>>>> Hi Paulo, >>>>> >>>>> some suggestion :) - could you provide full log (debug) and >>>>> core+binaries (for download) ? Or backtrace? >>>>> >>>>> regards, >>>>> bogdan >>>>> >>>>> Paulo Angonese wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> I have the same problem. >>>>>> When the SNOM ( 360 SIP 6.5.2 ) sends the REGISTER using TLS >>>>>> OpenSER crashes. >>>>>> I´m using >>>>>> OpenSer (1.1.1-tls (i386/linux)), >>>>>> openssl-0.9.7f-7.10 >>>>>> openssl-devel-0.9.7f-7.10 >>>>>> Fedora 2.6.11-1.1369_FC4 >>>>>> >>>>>> Any help? >>>>>> >>>>>> Thanks >>>>>> >>>>>> Paulo Angonese >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> [Users] CVS 31.03, TLS and dead tcp child >>>>>> Bogdan-Andrei Iancu bogdan at voice-system.ro >>>>>> Tue Oct 31 17:13:43 CET 2006 >>>>>> >>>>>> * Previous message: [Users] CVS 31.03, TLS and dead tcp child >>>>>> * Next message: [Users] Q:ABOUTE:SET UP MESSAGE FLOW >>>>>> * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] >>>>>> >>>>>> Hi Frank, >>>>>> >>>>>> can you pack the cores + binaries (lib + exe) + sources+logs(in >>>>>> debug >>>>>> mode) and put it somewhere for download? >>>>>> I will try to take a look. >>>>>> >>>>>> regards, >>>>>> Bogdan >>>>>> >>>>>> Frank DInnocenzo wrote: >>>>>> >>>>>> > Bogdan, >>>>>> > >>>>>> > I am using openser-1.1.0-tls_src. I actually get two core files >>>>>> after >>>>>> > the crash. Can I post them somehow? >>>>>> > >>>>>> > >>>>>> > # openser -V >>>>>> > version: openser 1.1.0-tls (i386/linux) >>>>>> > flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, >>>>>> > USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, >>>>>> > FAST_LOCK-ADAPTIVE_WAIT >>>>>> > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, >>>>>> MAX_LISTEN 16, >>>>>> > MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>>>> > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>>>>> > @(#) $Id: main.c,v 1.20 2006/07/04 17:25:54 bogdan_iancu Exp $ >>>>>> > main.c compiled on 14:27:50 Oct 24 2006 with gcc 4.0.0 >>>>>> > >>>>>> > Thanks >>>>>> > Frank >>>>>> > >>>>>> > >>>>>> > */Bogdan-Andrei Iancu <bogdan at voice-system.ro>/* wrote: >>>>>> > >>>>>> > Hi Frank, >>>>>> > >>>>>> > I'm successfully using a SNOM 320 with TLS and no problem so >>>>>> far..... >>>>>> > What openser version are you using?...do you get any core file >>>>>> > after the >>>>>> > crash?? >>>>>> > >>>>>> > regards, >>>>>> > bogdan >>>>>> > >>>>>> > Frank DInnocenzo wrote: >>>>>> > >>>>>> > > Hello, >>>>>> > > >>>>>> > > I'm new to openSer and wondered if there was any >>>>>> resolution to this >>>>>> > > post??? I'm hitting this same crash with openSer (with TLS >>>>>> enabled) >>>>>> > > using a Snom phone (300 with latest Beta firmware - >>>>>> > > snom300-6.5-SIP-j.bin ). I've gone >>>>>> > > through the archives looking for help getting this setup >>>>>> > working. I've >>>>>> > > also tried the Snom Softphone with the same result. >>>>>> > > >>>>>> > > Thanks >>>>>> > > Frank >>>>>> > > >>>>>> > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>> > > >>>>>> > > Hi, >>>>>> > > >>>>>> > > I have tried to register to OpenSER CVS 31.03 with my SNOM >>>>>> 360 >>>>>> > > (5.3.6sw), disabled pretty all modules from OpenSER (radius, >>>>>> > xlog, auth, >>>>>> > > ...) and always the server crashes. >>>>>> > > >>>>>> > > Is there something I have missed in configuration ? >>>>>> > > >>>>>> > > Thanks a lot for help, >>>>>> > > -Mika >>>>>> > > >>>>>> > > The error message and debug info seems like this: >>>>>> > > >>>>>> > > 11(24351) tcpconn_new: new tcp connection to: XXX.XX.XX.XXX >>>>>> > > 11(24351) tcpconn_new: on port 2171, type 3 >>>>>> > > 11(24351) tls_tcpconn_init: Entered: Creating a whole new ssl >>>>>> > connection >>>>>> > > 11(24351) tls_tcpconn_init: Looking up tls domain >>>>>> > [XXX.XX.XX.XXX:5061] >>>>>> > > 11(24351) tls_tcpconn_init: Using default tls server settings >>>>>> > > 11(24351) tls_tcpconn_init: Setting in ACCEPT mode (server) >>>>>> > > 11(24351) tcpconn_add: hashes: 442, 1 >>>>>> > > 11(24351) handle_new_connect: new connection: 0xb608f6a0 24 >>>>>> > flags: 0002 >>>>>> > > 11(24351) send2child: to tcp child 0 7(24347), 0xb608f6a0 >>>>>> > > 7(24347) received n=4 con=0xb608f6a0, fd=19 >>>>>> > > 7(24347) DBG: io_watch_add(0x810e580, 19, 2, 0xb608f6a0), >>>>>> fd_no=1 >>>>>> > > 7(24347) tls_update_fd: New fd is 19 >>>>>> > > 11(24351) DBG: handle_tcp_child: dead tcp child 0 (pid >>>>>> 24347, no 7) >>>>>> > > (shutting down?) >>>>>> > > 11(24351) DBG: io_watch_del (0x810e420, 17, -1, 0x0) >>>>>> fd_no=16 called >>>>>> > > 11(24351) ERROR: receive_fd: EOF on 15 >>>>>> > > 11(24351) DBG: handle_ser_child: dead child 7, pid 24347 >>>>>> > (shutting down?) >>>>>> > > 11(24351) DBG: io_watch_del (0x810e420, 15, -1, 0x0) >>>>>> fd_no=15 called >>>>>> > > 0(24339) child process 24347 exited by a signal 11 >>>>>> > > 0(24339) core was generated >>>>>> > > 0(24339) INFO: terminating due to SIGCHLD >>>>>> > > 6(24346) INFO: signal 15 received >>>>>> > > 6(24346) Memory status (pkg): >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)openser.org >>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>> >>>>> >>>>> >>>> >>>> >>> >>> >>> >> >> >

3 2

[Users] FreeBSD port update to 1.1.1 committed
by Jesus Rodriguez 01 Mar '07

01 Mar '07

Hello, For all that people asking me about the FreeBSD port version :) With a very long delay (sorry), today has been committed to the FreeBSD ports tree the update to OpenSER 1.1.1 version. I hope to have the update to 1.2 version ready and committed as soon as it is realesed. Regards, Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

1 0

[Serusers] RTPProxy codecs?
by Andrey Kuprianov 01 Mar '07

01 Mar '07

Hi, It just struck me. Does RTPProxy operate only with certain codecs or does it proxy RTP regardless the codec? I was just compiling RTPProxy code from CVS and saw this *********************************************************************************** * Please contact sales(a)sippysoft.com if you need G.729 support in makeann * *********************************************************************************** Regards, Andrey.

2 2

[Serusers] enum module error ser2.0
by tzieleniewski 01 Mar '07

01 Mar '07

Hi! I was trying to utilize the enum module of ser but whenever I try to do the check for the user part of the request uri I get the following error: ser@sen:~/ser/ser-cvs/proxy$ ./ser -c -f ser.cfg 0(5401) parse error (721,14-15): unknown command, missing loadmodule? I have the enum module loaded and the line for this happens is: if (is_user_from_e164()) { I also tried is_e164() which I found in the module header file but with the same result. it this is not going to work how can I make the check by hand? for instance having such uri sip:+4822102@domain.com will somthing like this work: uri=~ "sip:+[0-9]{7}@.*" ?? Bests Tomasz

2 2

Re[4]: [Users] OpenSER+ Freeradius+MySQL
by Ion Minzu 01 Mar '07

01 Mar '07

Hello raviprakash, 1. FreeRADIUS Version 1.1.3 2. I have: 1st PC 2nd PC OpenSER FreeRadius radiusclient-ng MySQL Am have radiusclient-ng. 3. I want total AAA. > Hello Ion, > Let me know, Which version using for freeradius > U want integrating the OpenSER + radius with MySQl > OpenSER nerver communicate with Radius, For that U need the RadiusClient-ng for that..... OpenSER--------->>radiusclient------>radius server > Let me know . What U want real with that...total AAA > On 3/1/07, Ion Minzu <ion.minzu(a)cts.md> wrote: > Salut, > I'm a beginner in radius, in the DB radius I have tables: nas, > radacct, radcheck, radgroupcheck, radgroupreply, radpostauth, > radreply, usergroup. I need to add users in DB (in the file > /etc/raddb/users is troublesome for me because I have 300 users > of the service). radacct is similar to table acc from openser, > but I don't understand what table from DB radius is similar to > table subscriber from DB openser. >> Salut, >>> Hello , >>> I have 2 servers: 1st is OpenSER, second is FreeRadius with MySQL. >>> Till now the authentication was making through mysql, but from now I >>> want make authentication and accounting through radius. In mysql are 2 >>> DB: radius and openser. >>> -To create a new user, in which DB it will be inserted? >>> >> you should add it in the DB used by Radius server >>> -Till now I add new users with openserctl, but with radius how to add >>> new users? >>> >> I'm not a radius user, but I guess you can use any DB tool, including >> mysql client ;). >> Regards, >> Bogdan > -- > Best regards, > Ion Minzu, > mailto:ion.minzu@cts.md > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

1 0

[Users] OpenSER+ Freeradius+MySQL
by Ion Minzu 01 Mar '07

01 Mar '07

Hello , I have 2 servers: 1st is OpenSER, second is FreeRadius with MySQL. Till now the authentication was making through mysql, but from now I want make authentication and accounting through radius. In mysql are 2 DB: radius and openser. -To create a new user, in which DB it will be inserted? -Till now I add new users with openserctl, but with radius how to add new users? -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

3 4

[Users] Multiple hops Accounting
by Manuel Guesdon 28 Feb '07

28 Feb '07

Hi, I try to have exploitable accounting for multiple domains and multi hops routing. I've previously tried for weeks on asterisk without success, even with adding c modules and playing with cdr fork :-( He is a case: - sip.domainA and sip.domainB point to the same Openser server. - 4(a)sip.domainA is dbaliased to 5(a)sip.domainB - 5(a)sip.domainB is dbaliased to 6@asteriskServer - 6@asteriskServer make a dial to phone2(a)sip.domainA I dial 4 on phone1 and get this INVITE routing phone1(a)sip.domainA === RURI:4@sip.domainA ==> openser openser === RURI:6@asteriskServer ==> asteriskServer asteriskServer == RURI:phone2@sip.domainA ==> openser openser ==> phone2 this works and phone2 ring, I take the call and hangup phone1 after some time. Now in acc table I find 15 records (7 INVITEs, 5 ACKs and 3 BYEs) The 3 BYE records seems correct (one from phone 1 to openser, one from openser to asteriskBox and the last from asteriskBox to phone 2 via openser). But I can't find a way to match the 7 INVITEs to the 3 BYEs, even with adding src&dst legs src_ip info and so on. >From acc records, I'd like to have make the following records report: == Call ID, src, dst, total duration, billing duration == - 1234, phone1@sip.domainA,4@sip.domainA,10,8 - 1234, 4@sip.domainA,5@sip.domainB,10,8 - 1234, 5@sip.domainB,6@asteriskBox,10,8 - 1234, 6@asteriskBox,phone2@domainA,10,8 Is it a complete dream or do you have an idea ? Manuel

2 1

[Serusers] problem in calling voicemail!!
by Meidiana Ten 28 Feb '07

28 Feb '07

Hi.. Thanx for the solution It really works well But now I face another problem Really need your help.. I configure SER for voicemail usage I'm using eyebeam as the UA. To check voicemail, I set it to call 113, and to send voicemail, it will call 114 based on the SER script. But it failed to forward to voicemail even the icon "send to voicemail" has appeared on the eyebeam. Below is my ser.cfg and ser_sems.cfg Please check it for me and tell me what's wrong. ser.cfg: debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) alias="pcr.ac.id" route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Offline Message Store Section # ----------------------------------------------------------------- if (is_from_local()) { if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted() && nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri =~ "sip:1.*@") { log("Forwarding to 5070"); rewritehostport("localhost:5070"); t_relay_to_udp("localhost","5070"); break; } if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.2:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } ser_sems.cfg: # ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # ser for sems configuration # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen=202.95.149.2 port=5070 children=4 #check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- alias="pcr.ac.id" # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # configure avpops db connection #modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) #modparam( "avpops", "avp_table", "subscriber" ) #modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database #modparam( "avpops", "db_scheme", # "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam("tm", "pass_provisional_replies", 1) # appends for INVITE to voicemail modparam("tm", "tw_append","voicemail_headers:hdr[User-Agent];P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[subject];msg[body]") # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); break; } # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if (!t_relay()) { sl_reply_error(); } break; }; # make transaction if (!t_newtran()){ log("could not create transaction\n"); sl_send_reply("500","could not create transaction"); break; } # actively absorb ACKs if (method == "ACK") { t_relay(); break; } # pass INFO to SEMS if (method=="INFO") { if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ log("could not contact sems\n"); t_reply("500","could not contact media server"); } } if (uri =~ "sip:101.*@") { if (!t_write_unix("/tmp/sems_sock","myapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:102.*@") { if (!t_write_unix("/tmp/sems_sock","myconfigurableapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:103.*@") { if (!t_write_unix("/tmp/sems_sock","myannounceapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:104.*@") { if (!t_write_unix("/tmp/sems_sock","myjukebox")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:105.*@") { if (!t_write_unix("/tmp/sems_sock","mycc")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:106.*@") { if (!t_write_unix("/tmp/sems_sock","ivr_announce")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } ############ default sems apps if (uri =~ "sip:110.*@") { if (!t_write_unix("/tmp/sems_sock","echo")){ log("could not contact echo\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:111.*@") { if (!t_write_unix("/tmp/sems_sock","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:112.*@") { if (!t_write_unix("/tmp/sems_sock","conference")){ log("could not contact conference\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:113.*@") { if (!t_write_unix("/tmp/sems_sock","mailbox")){ log("could not contact mailbox\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:114.*@") { if (!t_write_unix("/tmp/sems_sock","early_announce")){ log("could not contact early_announce\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("root@localhost","$email"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail/voicemail_headers")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); break; } break; } t_reply("404","Not found"); } Please tell me if you need my voicemail.conf..I'll send it to you. Thanx for the help Regards, Meidiana ____________________________________________________________________________________ We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265

2 1

[Users] DB migration from 1.1 to 1.2
by Bogdan-Andrei Iancu 28 Feb '07

28 Feb '07

Hi, today I uploaded on SVN trunk the new logic from migrating the existing DBs from 1.1 format to 1.2. The strategy is different than in the previous version. The old "reinstall" proved to be suboptimal because of the lack of flexibility and mainly it didn't provide a smooth transition (downtime was required in order to swap the DBs, unsafe - data could be lost if error occurred during reinstall - and data could be lost because of DB changes). A new approach was chosen this time - migration. A new DB is created in parallel with the old one and data is processed and transferred into the new DB via sql commands (and nor dumps), using strict mapping between old and new columns. This way the original data is kept and complex column renaming and data processing is possible (like lcr address for example). for the moment this migration is done only for mysql, but is possible to do it for postgres also but as I'm not to much into, I will need some help from some postgres guru ;). in the mean while testing is welcomed ;) regards, bogdan

1 0

[Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 28 Feb '07

28 Feb '07

---------- Forwarded message ---------- From: karim basraoui <basraouik(a)gmail.com> Date: 28 févr. 2007 20:33 Subject: Re: [Serusers] Call from Asterisk to SER to a foreign subscriber To: ram <talk2ram(a)gmail.com> I have found the thing! in Asterisk dial plan, when i'm trying to call my UA1, i should not do this: exten =>xxxx,Dial(SIP/UA1) but : exten =>xxxx,Dial(SIP/192.168.4.65/UA1) ;where 192.168.4.65 is the new host address thank you for your help ! 2007/2/24, karim basraoui <basraouik(a)gmail.com>: > > UA2 tries to call UA1 not UA3, sorry for the mistake > > 2007/2/24, karim basraoui < basraouik(a)gmail.com>: > > > > i had : > > > > UA1 (192.168.4.66:5061) > > > > }------------->Asterisk----------->SER-----------> UA3 ( > > 192.168.4.65:5061) > > UA2 ( 192.168.4.66:5062) > > > > then UA1 moved to SER domain : > > > > > > UA3 ( 192.168.4.65:5061) > > UA2 ( 192.168.4.66:5062) > > ------------->Asterisk----------->SER-----------> > > > > UA1 (192.168.4.66:5061) > > > > UA1 registers to SER and SER forwards the registration to ASTERISK, > > Asterisk is aware of the new location. > > > > UA2 tries to call UA3. It sends the invite to Asterisk and Asterisk > > replies with a TRYING message. > > after a while the call fails with an error > > > > > > 2007/2/24, ram < talk2ram(a)gmail.com>: > > > > > > then register with SER, should do the job > > > > > > UA----ASterisk----SER-----Phone > > > > > > is this correct > > > > > > ram > > > > > > > > > On 2/24/07, karim basraoui <basraouik(a)gmail.com > wrote: > > > > > > > > Hi, > > > > > > > > the call i'm trying to establish is from ASTERISK to SER. > > > > the UA i'm calling was initially in ASTERISK domain and moved to SER > > > > domain , a roaming case. > > > > > > > > 2007/2/24, ram < talk2ram(a)gmail.com>: > > > > > > > > > > Hi > > > > > > > > > > your question is not clear > > > > > > > > > > how is the call path happening from ASterisk or from SER > > > > > or both are gateway proxies ? > > > > > > > > > > ram > > > > > > > > > > > > > > > On 2/23/07, karim basraoui < basraouik(a)gmail.com > wrote: > > > > > > > > > > > Hi, > > > > > > > > > > > > I've set up a small architecture in which an ASterisk proxy is > > > > > > connected to a SER proxy in two different domains. > > > > > > > > > > > > I have sip phones connected to both of them. when an asterisk > > > > > > subscriber "travels" to SER domain it tries to register to SER. I've > > > > > > configured SER to accept foreign > > > > > > subscribers and to REGISTER them to their initial domain by the > > > > > > command "t_replicate". Till now everything works good, the UA is registered > > > > > > and Asterisk is aware > > > > > > of its new location, it displays in peers list the UA and its > > > > > > new IP address and port. > > > > > > > > > > > > The problem is when i'm trying to call the foreign UA from an > > > > > > Asterisk subscriber, it doesn't work. i have a TRYING response but no call > > > > > > establishement. > > > > > > > > > > > > how can i overcome this problem please? > > > > > > > > > > > > -- > > > > > > BASRAOUI Karim > > > > > > www.basraouik.afrikart.net > > > > > > _______________________________________________ > > > > > > Serusers mailing list > > > > > > Serusers(a)lists.iptel.org > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > BASRAOUI Karim > > > > www.basraouik.afrikart.net > > > > > > > > > > > > > > > > -- > > BASRAOUI Karim > > www.basraouik.afrikart.net > > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net > -- BASRAOUI Karim www.basraouik.afrikart.net -- BASRAOUI Karim www.basraouik.afrikart.net

1 0

[Users] New OpenSER Config Generator features
by Andreas Granig 28 Feb '07

28 Feb '07

Hi all, Three new features for sip:wizard, the OpenSER Config Generator, have been released this week: - user-aliases: You can now assign aliases to local users. This is especially useful for mapping numeric numbers to SIP accounts for PSTN inward dialing. - cli-pai: This feature allows the provisioning of arbitrary SIP URIs, which are inserted into a P-Asserted-Identity Header for outgoing calls. This for example allows the signalling of a numeric CLI to a PSTN gateway (so it’s the opposite of user-aliases). - ring-timeout: You can now dynamically set the ring-timeout (or, more specifically, the fr_inv_timer) per local callee. Visit http://www.sipwise.com/wizard to give it a try. Regards, Andreas

3 6

Re: [Users] 3xx requests and t_relay
by Bogdan-Andrei Iancu 28 Feb '07

28 Feb '07

Hi Shane, have you followed this ex: http://www.openser.org/docs/modules/1.2.x/uac_redirect.html#AEN251 can you be more specific about the errors you see? regards, bogdan Shane Burrell wrote: > I am calling the t_relay after getting the contacts. However internally I > see in openser it makes it back to the route and then dies in loose route. > I assumed there was something I need to check to allow the new call to be > processed. > > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, February 28, 2007 12:26 PM > To: Shane Burrell > Cc: users(a)openser.org > Subject: Re: [Users] 3xx requests and t_relay > > Hi, > > let me see if I get it right...you get a 3xx reply and try to generate > INVITE to the new contacts? if so, take a look at the uac_redirect module. > > regards, > bogdan > > Shane Burrell wrote: > >> I have this working to the point it tries to relay the call but I see >> that internally openser is not making the new call. I assume t_relay >> calls a new route but how to I make sure that all relays get >> forwarded. Some of my logic else where seems to be killing the >> rerouted call and I need any 302 routed calls to be accepted and routed. >> >> >> >> TIA, >> >> >> >> Shane >> >> >> *************************************************************** >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

1 0

[Users] 3xx requests and t_relay
by Shane Burrell 28 Feb '07

28 Feb '07

I have this working to the point it tries to relay the call but I see that internally openser is not making the new call. I assume t_relay calls a new route but how to I make sure that all relays get forwarded. Some of my logic else where seems to be killing the rerouted call and I need any 302 routed calls to be accepted and routed. TIA, Shane ***************************************************************

2 1

RE: [Users] cross nat pinging
by Marcus Hunger 28 Feb '07

28 Feb '07

Hi Fatih, you could spoof the natping's source-ip so it would apear to come from the same host. I created a patch for that. It uses raw-sockets and works for me on linux. The patch is made against trunk. To use it, just set the udpping_from parameter of module nathelper. modparam("nathelper", "udpping_from" , "myhost:myport") Feedback on this would be welcome. Best regards, Marcus Fatih Cerit wrote: > Hi all > > I am testing nat pinging. I have a natted UA and 2 opensers with > diffrent ips. I decided to ping same nat from 2 opensers. I am going to > have ability to increase ping interval from 15 seconds to 30 > seconds. But I don't have any idea that cross pinging works or not. Is > there any one at there whom tested before or any suggestions?? > > Thanks > Fatih Cerit -- Marcus Hunger - hunger(a)sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 133/5838/1589, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 1

[Users] AVPPOPS preformace question (OpenSER 1.1.x)
by Mike O'Connor 28 Feb '07

28 Feb '07

Hi Guys What is going to be faster: if (avp_db_load("$ruri/username", "$avp(s:callfwdimmediate)")) { .... }; if (avp_db_load("$ruri/username", "$avp(s:callfwdbusy)")) { .... }; if (avp_db_load("$ruri/username", "$avp(s:callfwdnoanswer)")) { .... }; or (if it works): 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)")) { .... }; If the second was valid I would have thought it would be quicker. Thanks Mike

2 1

[Serusers] register_watcher(): Domain 'location' not found
by SER LIST 28 Feb '07

28 Feb '07

Hello, I am getting the message register_watcher(): Domain 'location' not found logged in my /var/log/messages file. Any idea what this is and to what it could be pointing to. I am running ser-2.0 daily-snapshop from Feb24. Thanks in advance _________________________________________________________________ Windows Live Spaces: share your New Year pictures! http://spaces.live.com/?mkt=en-ca

1 0

[Users] SIP Firewall
by Sergio G 28 Feb '07

28 Feb '07

Hello, I'm Sergio... Can anybody tell me if OpenSer could be used in the following way: ANY SIP REQUEST != REGISTER (to port 5060) ----------------------------> OpenSER --------> SOME PROXY_SIP (To port 5060) and the REGISTERs Request, redirected to another PROXY SIP and to another port... REGISTER REQUEST (to port 5060) --------------------------------> OPENSER------> REGISTRAR PROXY (to port 5062) I need to use it as some kind of Firewall SIP, REGISTER redirected to some IP and to some port, and the other requests redirected to another IP and port diferent. If this could be done, How? Where should I start reading? Thanks any help, Sergio

2 1

[Users] Multiple rtpproxy
by Ibrahim Hamouda 28 Feb '07

28 Feb '07

Hi Guys Could anybody give me a code snippet how to setup multiple rtpproxy in openser.cfg Thanks Ibrahim Hamouda

2 1

[Users] cancelling INVITE during OK response
by Web Monkey 28 Feb '07

28 Feb '07

hi, is there a way to cancel or terminate an INVITE or call upon receipt of the OK reply. my configuration invokes an external program which either return success or failure for OK message (response to INVITE). when failure is returned, i want the script to terminate the call or cancel the INVITE (OK message will not be forwarded to the calling UA). is there an existing function or module to realize this? the scenario is drawn below with provisional messages removed. UAC1 ------------------ proxy1 ----------------- proxy2 --------------- UAC2 call: INVITE ---------------------------> INVITE ----------------------> INVITE -----------------> OK <---------------- *XXX - error occurred during processing OK message thanks, sho --------------------------------- It's here! Your new message! Get new email alerts with the free Yahoo! Toolbar.

2 1

[Users] openserctl problem
by Helmut Kuper 28 Feb '07

28 Feb '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, After fixing that Query error I tried to add 2 user to openser. The first was successfully added to database. the 2nd failed with: ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 ERROR: introducing the new user '103(a)sip0.en.ewetel.de' to the database failed I looked into the database table subscribers and saw that column phplib_id is of type UNIQUE. I looked into the detailed mysql.log file and find out that the executed INSERT statement doesn't set the column phplib_id, so the 2nd user would have the same phplib_id value ('' or NULL) as the first. Because of column typ UNIQUE this will lead to that error above. Maybe someone of the developers can fix this. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5Etw4tZeNddg3dwRAiX3AJwKD0DDVwKDxbeX1/9fVitIxQSaKgCgliv5 u5IIkongIw3a16e4uu+vm4A= =kV49 -----END PGP SIGNATURE-----

2 1

[Users] openserctl error
by Helmut Kuper 28 Feb '07

28 Feb '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, the openserctl of the latest SVN trunk seems to has an error in the user add command. Line 1544 looks like this: UERY="insert into $SUB_TABLE ($ instead of QUERY="insert into $SUB_TABLE ($ regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5EdQ4tZeNddg3dwRAr7xAKCjGVi++E6eX3EhDVOj3Bm85FHSZACfd3OA 6xeUy4rZ7urCebfmALAjApQ= =93G4 -----END PGP SIGNATURE-----

2 1

[Users] OpenSER wiki semi-dead?
by Tais M. Hansen 28 Feb '07

28 Feb '07

Hi, No matter what subpage I try to access on the wiki, only the front page is displayed. Is anyone else seeing this? http://openser.org/dokuwiki/doku.php -- Regards, Tais M. Hansen ___________________________________________________________ http://www.linkedin.com/in/thansen

2 2

[Users] Auth with Radius+Kerberos
by Stefano Capitanio 28 Feb '07

28 Feb '07

Hi List, we are trying to do authentication with SER -> RADIUS -> KERBEROS following draft-kaushik-radius-sec-ext (Radius Secutity Extensions Using Kerberos) has anyone succeeded in doing this (or something similar)? thanks, Stefano

2 2

[Users] OpenSER 1.2.0
by Danilo Vilela Dantas 28 Feb '07

28 Feb '07

Hi all, We are working VoIP Project. We are using OpenSER 1.0.1. Unfortunately it doesn't support xcap and presence modules. Only the version 1.2.0 of the OpenSER. Is it ready openser 1.2.0? Or still delay? Thanks, regards, Danilo

4 3

[Users] cross nat pinging
by Fatih Cerit 28 Feb '07

28 Feb '07

Hi all I am testing nat pinging. I have a natted UA and 2 opensers with diffrent ips. I decided to ping same nat from 2 opensers. I am going to have ability to increase ping interval from 15 seconds to 30 seconds. But I don't have any idea that cross pinging works or not. Is there any one at there whom tested before or any suggestions?? Thanks Fatih Cerit

2 1

[Serusers] ser usrloc expiration
by Corey 28 Feb '07

28 Feb '07

The command "serctl ul show" displays an 'Expires' entry. I'm wondering: * what is the default expiration value? * where/how does this get set, and is it configurable? Many thanks!

2 1

[Users] TLS between UA and Openser
by Jayesh Nambiar 28 Feb '07

28 Feb '07

Hi everybody, I am trying to achieve TLS setup between the UAs and OpenSER. I am using PAP2 and SPA series for testing. Now the PAP2 does not support TLS transport, is it possible to use TLS setup between OpenSER and PAP2 using DNS-SRV? I tried using DNS-SRV but was not successful. The SPA supports TLS transport. I have generated the priviate key and required certs in the OpenSER. For SPA to work, how do we incorporate the certificate inside the SPA? Or should I get a certificate generated from Linksys and then add it to my CA list in OpenSER? I am not very familiar with the TLS concept but still learning. Thanks in advance. w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

1 0

[Users] How do to prefix or pattern based routing
by Chris Heiser 28 Feb '07

28 Feb '07

It's more than possible there's a module or piece of documentation that I've been unable to uncover in my attempts to solve this issue. Basically, I'm looking for a way to do prefix or pattern based routing using a datasource (such as dbtext or a real database). I don't want to have to hack an openser.cnf file for each server I setup. That would just be silly. I've looked at PDT, but 1) it does source domain matching, which I could really care less about, and 2) it's misbehaving or misdocumented, as it won't allow for more than one prefix to be defined for a single source domain. I'd really love to be able to do pattern matches and route that way. But at this point, anything is better than hacking the config file. Any suggestions? --Chris

2 1

[Users] new MediaProxy release 1.8.2
by Adrian Georgescu 27 Feb '07

27 Feb '07

Hello, There is a new MediaProxy release available. To upgrade go to http://mediaproxy.ag-projects.com or download the software directly from: http://mediaproxy.ag-projects.com/mediaproxy-1.8.2.tar.gz Changes from version 1.8.1 to 1.8.2 ----------------------------------- - Added support to setup sessions by either caller or called as the initial INVITE may not contain a SDP body (Jeff Williams <jeffw(a)globaldial.com>) - Added support to update the RTP stream sign-in addresses after re- INVITES (Jeff Williams <jeffw(a)globaldial.com>) Changes from version 1.8.0 to 1.8.1 ----------------------------------- - Added User-Name attribute to the radius accounting Update request (freeradius needs it to be able to proxy the request) This release includes new functionality. Your feedback is appreciated, if you wish to provide it please reply to the mailing lists. Kind regards, Adrian Georgescu

3 8

[Users] web gui for registered openser user
by alper ozbilen 27 Feb '07

27 Feb '07

Dear All, I am new in openser. I have just installed openser on Fedora core 4 and run it. *I need "php user gui" for user to make web-based registeration process. I will able to bult it up but i dont know which mysql table i should you. *My aim is to give service to registered user on my datebase. In order to implement it, which open module should i install ? any help wolud be appreciated Thank a lot with my best regard. Alper Ozbilen _________________________________________________________________ Want a degree but can't afford to quit? Top school degrees online - in as fast as 1 year http://forms.nextag.com/goto.jsp?url=/serv/main/buyer/education.jsp?doSearc…

3 4

[Serusers] ser2 and sems?
by TZieleniewski 27 Feb '07

27 Feb '07

Hi!! I am going to set up the sems-ser configuration. In the mini-sems-ser-howto there is a clue to have separate instance of ser for sems. Will sems 0.10.0 work with the ser ottendorf release?? Bests tomasz

1 0

[Users] install:1.1.x-to-1.2.x
by Daniel-Constantin Mierla 27 Feb '07

27 Feb '07

(started a new thread from a devel discuttion) Everybody is encouraged to update the page below: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x If you do not know how to explain clear, but you know (feel) something changed from 1.1.x to 1.2.0 in a module or other place, just introduce in bold red a short phrase at the beginning or in module's section. Someone will see later and elaborate it. Cheers, Daniel On 02/27/07 13:01, Andreas Granig wrote: > Thank god there is > http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x > > Hopefully, all the small changes are documented there (hint: the one > below isn't yet) ;o) > > Cheers, > Andreas > > Bogdan-Andrei Iancu wrote: >> Hi Juha, >> >> yes, there is an impact on the script, but very small. The only >> change is: >> modparam("registrar", "sip_natping_flag", x ) >> goes into >> modparam("nathelper","sipping_bflag",x) >> >> regards, >> bogdan

1 0

[Users] Presence module
by Helmut Kuper 27 Feb '07

27 Feb '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I just configured the presence modul and got some strange output in my shell when my xlite phone trys to subscribe to a non existant user: 0(16615) ERROR: warning_builder: buffer size exceeded 0(16615) WARNING: warning skipped -- too big 0(16615) ERROR: warning_builder: buffer size exceeded 0(16615) WARNING: warning skipped -- too big 0(16615) ERROR: warning_builder: buffer size exceeded 0(16615) WARNING: warning skipped -- too big 0(16615) PRESENCE: get_subs_dialog:The query for subscribtion for [user]= sebastian.bruening%40googlemail.comtalk.v104112f54d0,[domain]= sip0.en.ewetel.de for [event]= presence.winfo returned no result 0(16615) PRESENCE:query_db_notify: Could not get subs_dialog from database 0(16615) PRESENCE:update_subscribtion:Could not send notify for presence.winfo 0(16615) PRESENCE:get_xcap_tree:The query in table xcap for [username]=sebastian.bruening%40googlemail.comtalk.v104112f54d0 , domain=sip0.en.ewetel.de returned no result any ideas what this mean? I use latest version of openser (1.2.0). regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5D+D4tZeNddg3dwRAiYcAKCt9FJYu/yM6+HW/+QOXlp0XOAPWACfYpFZ 0JbJo+z27a2M1zr8WlEG4G4= =I9/o -----END PGP SIGNATURE-----

2 1

[Users] bug in radiusclient ng for 64b
by Stuart Clark 27 Feb '07

27 Feb '07

Hello, I have a 64bit machine running openser 1.1.0, and wish to log radius accounting details to radius using the acc module and radiusclient-ng. I am facing issues talking to a freeradius server, as described at: https://developer.berlios.de/bugs/?func=detailbug&bug_id=7442&group_id=1208 Basically, the server reports: Error: Received Accounting-Request packet from w.x.y.z with invalid signature! (Shared secret is incorrect.) I have updated to the cvs version of radiusclient-ng, but the problem still exists. It would appear that the md5 routine is not compatible with the md5 routine used by the freeradius software. Just wondering is anyone else has worked their way around this problem. Cheers, Stuart > X-Archived: http://article.gmane.org/gmane.comp.voip.openser.user/4415 > From: Daniel-Constantin Mierla <daniel@...> > Subject: FYI: bug in radiusclient ng for 64b > Newsgroups: gmane.comp.voip.openser.user, gmane.comp.voip.openser.devel > Date: 2006-05-16 01:44:16 GMT (13 weeks, 3 days and 58 minutes ago) > > On 05/16/06 02:48, Maxim Sobolev wrote: >> Hi, >> >> Reportedly there is an problem in radiusclient library with md5 hash >> calculation on 64bit platforms (amd64, sparc64, etc), which results in >> server rejecting requests even with proper secret configured on >> client. I have applied a fix, but don't have access to any 64bit >> hardware, therefore can't verify. >> >> Therefore, if anybody is having problems with radiusclient on 64bit >> arch he is more than welcome to try update to the latest version from >> cvs and let me know if it helps or not. Instructions on checking out >> sources from berlios cvs are here: >> https://developer.berlios.de/cvs/?group_id=1208. >> >> Thanks! >> >> -Maxim >

3 2

[Users] OpenSER administrator issue
by Pat wang 27 Feb '07

27 Feb '07

Hi, Just managed to install the new openser administrator and everything just works. Thanks to Mike for this nice application. However I can't add user with openserctl anymore. The following error appears: ERROR 1054 (42S22) at line 1: Unknown column 'phplib_id' in 'field list' ERROR: introducing the new user '5702' to the database failed On the other hand, adding new users from administrator GUI does not have this problem. This happened right after installation of aministrator so I think it is related to the administrator mysql table changes. Any suggestions please? Thanks, Patrick

4 4

[Users] eta for v1.2.0 - March 12
by Daniel-Constantin Mierla 27 Feb '07

27 Feb '07

Hello everybody, just to give you an update with estimation time line for release 1.2.0... The plan is to release it on the 12th of March, hard work being with packaging files update, migration tools, and of course still with testing and stress-testing ... If you have good DB (mysql/postgres) skills and want to help, just say it, it is enough work around :-) . Some tables changed the structure of the fields (like LCR with the IP address), and a bit more advanced migration mechanism should be in place. Of course, help with testing is always appreciated ... Cheers, Daniel

1 0

[Serusers] PATH Module in SER?
by SER LIST 27 Feb '07

27 Feb '07

Is there a paths module for SER? I want to implement a Load balancer using Dispatcher module and I will need a path module for handling registrations. Thanks in advance _________________________________________________________________ Your Space. Your Friends. Your Stories. Share your world with Windows Live Spaces. http://spaces.live.com/?mkt=en-ca

1 0

[Users] OpenSER CVS was migrated to SVN
by Daniel-Constantin Mierla 27 Feb '07

27 Feb '07

Hello, after the discussion on devel mailing list was decided to migrate OpenSER CVS repository to SVN. Most of developers considered the SVN more convenient to work with, especially when hunting reported bugs (svn revision number is printed by openser -V). Unfortunately this bring a bit of work for those installing and maintaining OpenSER from CVS. The have to grab the subversion (svn) tree righ now. There are no changes in directory structure, so any patch you had privately should apply easy. To get the same directory tree as for CVS: # svn co https://openser.svn.sourceforge.net/svnroot/openser sip-server For details about how to get a certain branch, please see the download page on http://www.openser.org: http://www.openser.org/index.php?option=com_content&task=view&id=29&Itemid=… We are still working on making snapshot system using SVN, and maybe some other tools are not yet updated, but they will be in the next days. However, what is most important should be ready to go -- source tree should be fully functional with SVN. Please let us know if you encounter any issue. The translation to SVN was made over this weeken. New commits were already done to SVN. There was no critical update you should run after, but is recommended to get SVN version as soon as possible. Old branches were migrated (releases 0.9.x, 1.0.x, 1.1.x). For the reference, initial revision for subversion is 1689. Once you get your patch against latest CVS version availabe, then should be no issue to apply it to this subversion revision. Full commit history from CVS is available on SVN. Hope SVN will make openser maintenance easier for all of you. Here are some links to get you familiar with SVN commands and capabilities: http://svnbook.red-bean.com/nightly/en/index.html http://subversion.tigris.org/ OpenSER SVN page at sourceforge: http://sourceforge.net/svn/?group_id=139143 Cheers, Daniel

4 6

Re: [Users] dialog module configuration question
by Bogdan-Andrei Iancu 26 Feb '07

26 Feb '07

Hi Andy, yes, you are correct - the package 7 (the 200 OK) must mirror the Record-Route set from the request. For this you need to enable the rrs param : <recv request="INVITE" crlf="true" rrs="true"> </recv> regards, bogdan Andy Pyles wrote: > Hi Bogdan, > > ok let me go back to my example: > > Here's more detail: > 192.168.0.101 = Caller (sipp uas) > 1.2.3.4 = openser > 4.3.2.1 = callee ( sipp uac) > > > 1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request: INVITE > sip:service@1.2.3.4:5060, with session description > 2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving a try > 3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE > sip:service@4.3.2.1:5060, with session description > 4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180 Ringing > 5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200 OK, with session > description > 6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180 Ringing > 7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200 OK, with session > description > 8.) 192.168.0.101 -> 1.2.3.4 SIP Request: ACK > sip:service@1.2.3.4:5060 > 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK sip:service@4.3.2.1:5060 > 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE > sip:service@1.2.3.4:5060 > 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE sip:service@4.3.2.1:5060 > 12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK > 13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK > > So, you are saying for Packets 8, 10 I should add the '[routes]' logic > to sipp. How this works is: from the sipp documentation: "rrs: Record > Route Set. if this attribute is set to "true", then the > "Record-Route:" header of the message received is stored and can be > recalled using the [routes] keyword.". > > This I completey agree with. sipp Must be sending the Route: header > in Packets 8 and 10. However, packet 7 MUST have the Record-route > header, otherwise, How can sipp can put the correct value into the > Route: header. See my point? > > Reference: rfc 3665 ( secion 3.2 Packet f11, f14) > > regards, > Andy > > > On 2/23/07, Andy Pyles <andy.pyles(a)gmail.com> wrote: >> Hi Bogdan, >> >> correct. but on client config "[routes]" ( for sipp) will only work >> IF the client receives a Record-route. Since I'm not, it doesn't help >> me. Am I missing something? >> >> Andy >> >> On 2/23/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: >> > Hi Andy, >> > >> > in client config, you need to add "[routes]" for ACK and BYE messages >> > (take a look at the cfg I sent you) >> > >> > regards, >> > bogdan >> > >> > Andy Pyles wrote: >> > > I Just re-read the docs on loose_route(). So please disregard this >> > > question. ( only processed if Route: header is present. Which isn't >> > > present because Record-route: header isn't being sent to caller ) >> > > >> > > So, I'm still trying to figure out why record-route: header is not >> > > being sent to caller. >> > > >> > > >> > > On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote: >> > >> Hi Bogdan, >> > >> >> > >> After running additional debugs, for some reason the call to >> > >> loose_route() is failing. >> > >> >> > >> if (loose_route()) { >> > >> # mark routing logic in request >> > >> xlog("L_INFO", "loose_route() succeeded\n "); >> > >> route(1); >> > >> } else{ >> > >> xlog("L_INFO", "loose_route()failed - M=$rm RURI=$ru F=$fu >> > >> T=$tu IP=$si ID=$ci\n"); >> > >> }; >> > >> >> > >> >> > >> Any ideas why this could be occuring? >> > >> >> > >> >> > >> On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote: >> > >> > HI Bogdan, >> > >> > >> > >> > I'm already using an almsot identical version of uas.xml and >> uac.xml ( >> > >> > yes rrs=true) is being used. However in your version the uas.xml >> > >> > doesn't have rrs="true" after initial invite which I think is >> needed. >> > >> > See as you can see below, setting rrs="true" for uac will only >> work if >> > >> > it receives a Record-Route header in the 200OK which it's not. >> > >> > >> > >> > In this case, ALL messages from openser to sipp uac do not >> contain the >> > >> > Record-route header. So I don't think it's a sipp problem, but an >> > >> > openser configuration problem. I've tried using other devices >> for a >> > >> > uac, such as x-lite but the same problem. >> > >> > >> > >> > Andy >> > >> > >> > >> > On 2/22/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: >> > >> > > Hi Andy, >> > >> > > >> > >> > > so it's about sipp :D - I remember I had some hard times to >> make >> > >> it work >> > >> > > with record Route. >> > >> > > >> > >> > > take a look at the attached files, they might help you. >> > >> > > >> > >> > > regards, >> > >> > > bogdan >> > >> > > >> > >> > > Andy Pyles wrote: >> > >> > > > HI Bogdan, >> > >> > > > >> > >> > > > thanks for your reply. >> > >> > > > yes you are correct. The Bye doesn't have the Route header. >> > >> > > > It appears the the 200 OK sent to the caller doesn't >> contain a >> > >> > > > Record-route header. >> > >> > > > Messages between openser and callee contain record-route >> > >> information, >> > >> > > > but messages between caller and openser do not. >> > >> > > > Is there a way to enable that? >> > >> > > > >> > >> > > > Here's more detail: >> > >> > > > 192.168.0.101 = Caller (sipp) >> > >> > > > 1.2.3.4 = openser >> > >> > > > 4.3.2.1 = callee ( sipp) >> > >> > > > >> > >> > > > >> > >> > > > 1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request: INVITE >> > >> > > > sip:service@1.2.3.4:5060, with session description >> > >> > > > 2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving a try >> > >> > > > 3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE >> > >> > > > sip:service@4.3.2.1:5060, with session description >> > >> > > > 4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180 Ringing >> > >> > > > 5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200 OK, with >> > >> session >> > >> > > > description >> > >> > > > 6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180 Ringing >> > >> > > > 7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200 OK, with >> > >> session >> > >> > > > description >> > >> > > > 8.) 192.168.0.101 -> 1.2.3.4 SIP Request: ACK >> > >> > > > sip:service@1.2.3.4:5060 >> > >> > > > 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK >> > >> sip:service@4.3.2.1:5060 >> > >> > > > 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE >> > >> > > > sip:service@1.2.3.4:5060 >> > >> > > > 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE >> > >> sip:service@4.3.2.1:5060 >> > >> > > > 12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK >> > >> > > > 13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK >> > >> > > > >> > >> > > > --- >> > >> > > > Packets 6,7 and following contain no Record-route >> information. >> > >> > > > The other weird thing is that openser is passing on the >> Route: >> > >> header >> > >> > > > it recevied from callee to the caller. >> > >> > > > >> > >> > > > >> > >> > > > Please see attached for complete ngrep output. >> > >> > > > >> > >> > > > >> > >> > > > On 2/21/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> >> wrote: >> > >> > > >> Hi Andy, >> > >> > > >> >> > >> > > >> could you check on the net if the BYE contain the Route hdr >> > >> added to >> > >> > > >> INVITE as Record-Route? I have some doubts on this as I see: >> > >> > > >> 0(966) find_first_route: No Route headers found >> > >> > > >> 0(966) loose_route: There is no Route HF >> > >> > > >> >> > >> > > >> and if the BYE is not identified, the dialog is not closed. >> > >> > > >> >> > >> > > >> regards, >> > >> > > >> bogdan >> > >> > > >> >> > >> > > >> Andy Pyles wrote: >> > >> > > >> > 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 >> > >> > > >> > >> > >> >

2 3

[Users] Exported statistics and pseudo-variables
by Michel Bensoussan 26 Feb '07

26 Feb '07

Hello I didn't understand how to use the exported statistics and the pseudo-variables. In module dialog, active_dialogs is an exported statistic and $dlg_count a pseudo variable. How can I get them in code and in cfg file? Can I write in cfg file something like: if ($dlg_count > 10){...}? (not in this way of course but that's the idea). Thanks, Michel. PS: I didn't understand the example in http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.1.x -- Michel Bensoussan Extricom Ltd. Tel-Aviv University, The Wolfson Computer and Software Engineering Building 30 Haim Levanon Street, P.O.B. 39040 Tel-Aviv, 61390 Israel Tel: +972 3 745 1000 ext 1065 Fax: +972 3 745 1001

2 5

[Users] Adding openser to an already in use asterisk pbx
by Ben Ferguson 26 Feb '07

26 Feb '07

Hello all, I am very new to openSER and have been going over MANY, MANY pages of info on openSER (and SER) and am having a bit of a problem figuring out what exactly i need to do for what I am trying to have happen. Basically it's like this: I have a small company running a NATed asterisk pbx (there is actually about 30 to 40 users already setup and using Asterisk). Now I would like to add openSER to allow some outsourced/remote users via voip so they may dial anyone in the company directly over their voip connection. Basically, all asterisk users can call all asterisk users and also call remote (openSER) users; and all remote (openSER) users can call Asterisk users. I know Asterisk decently well, but I am not sure what I would need to set up in openSER to handle this. My Asterisk box is currently not set up for DB auth. I just have the users set up in sip.conf. Re-doing the Asterisk box is absolutely out of the scope of this project. I would just like to set up some remote users via openSER. (I do have a public IP set up for my openSER box.) Anyone have this type of setup going? Or anyone can tell me how this would best be done? (I'd love to see an openser.cfg of this setup....) Thanks in advance, Ben Jeffries fjsixty [at] gmail [dot] com

1 0

[Users] TLS and dead tcp child
by Paulo Angonese 26 Feb '07

26 Feb '07

Hi, I have the same problem. When the SNOM ( 360 SIP 6.5.2 ) sends the REGISTER using TLS OpenSER crashes. I´m using OpenSer (1.1.1-tls (i386/linux)), openssl-0.9.7f-7.10 openssl-devel-0.9.7f-7.10 Fedora 2.6.11-1.1369_FC4 Any help? Thanks Paulo Angonese [Users] CVS 31.03, TLS and dead tcp child Bogdan-Andrei Iancu bogdan at voice-system.ro Tue Oct 31 17:13:43 CET 2006 * Previous message: [Users] CVS 31.03, TLS and dead tcp child * Next message: [Users] Q:ABOUTE:SET UP MESSAGE FLOW * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi Frank, can you pack the cores + binaries (lib + exe) + sources+logs(in debug mode) and put it somewhere for download? I will try to take a look. regards, Bogdan Frank DInnocenzo wrote: > Bogdan, > > I am using openser-1.1.0-tls_src. I actually get two core files after > the crash. Can I post them somehow? > > > # openser -V > version: openser 1.1.0-tls (i386/linux) > flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, > USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, > FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > @(#) $Id: main.c,v 1.20 2006/07/04 17:25:54 bogdan_iancu Exp $ > main.c compiled on 14:27:50 Oct 24 2006 with gcc 4.0.0 > > Thanks > Frank > > > */Bogdan-Andrei Iancu <bogdan at voice-system.ro>/* wrote: > > Hi Frank, > > I'm successfully using a SNOM 320 with TLS and no problem so far..... > What openser version are you using?...do you get any core file > after the > crash?? > > regards, > bogdan > > Frank DInnocenzo wrote: > > > Hello, > > > > I'm new to openSer and wondered if there was any resolution to this > > post??? I'm hitting this same crash with openSer (with TLS enabled) > > using a Snom phone (300 with latest Beta firmware - > > snom300-6.5-SIP-j.bin ). I've gone > > through the archives looking for help getting this setup > working. I've > > also tried the Snom Softphone with the same result. > > > > Thanks > > Frank > > > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> > > > > Hi, > > > > I have tried to register to OpenSER CVS 31.03 with my SNOM 360 > > (5.3.6sw), disabled pretty all modules from OpenSER (radius, > xlog, auth, > > ...) and always the server crashes. > > > > Is there something I have missed in configuration ? > > > > Thanks a lot for help, > > -Mika > > > > The error message and debug info seems like this: > > > > 11(24351) tcpconn_new: new tcp connection to: XXX.XX.XX.XXX > > 11(24351) tcpconn_new: on port 2171, type 3 > > 11(24351) tls_tcpconn_init: Entered: Creating a whole new ssl > connection > > 11(24351) tls_tcpconn_init: Looking up tls domain > [XXX.XX.XX.XXX:5061] > > 11(24351) tls_tcpconn_init: Using default tls server settings > > 11(24351) tls_tcpconn_init: Setting in ACCEPT mode (server) > > 11(24351) tcpconn_add: hashes: 442, 1 > > 11(24351) handle_new_connect: new connection: 0xb608f6a0 24 > flags: 0002 > > 11(24351) send2child: to tcp child 0 7(24347), 0xb608f6a0 > > 7(24347) received n=4 con=0xb608f6a0, fd=19 > > 7(24347) DBG: io_watch_add(0x810e580, 19, 2, 0xb608f6a0), fd_no=1 > > 7(24347) tls_update_fd: New fd is 19 > > 11(24351) DBG: handle_tcp_child: dead tcp child 0 (pid 24347, no 7) > > (shutting down?) > > 11(24351) DBG: io_watch_del (0x810e420, 17, -1, 0x0) fd_no=16 called > > 11(24351) ERROR: receive_fd: EOF on 15 > > 11(24351) DBG: handle_ser_child: dead child 7, pid 24347 > (shutting down?) > > 11(24351) DBG: io_watch_del (0x810e420, 15, -1, 0x0) fd_no=15 called > > 0(24339) child process 24347 exited by a signal 11 > > 0(24339) core was generated > > 0(24339) INFO: terminating due to SIGCHLD > > 6(24346) INFO: signal 15 received > > 6(24346) Memory status (pkg):

2 3

[Users] FOSDEM 2007 - openser slides
by Daniel-Constantin Mierla 26 Feb '07

26 Feb '07

Hello, if anybody interested in the slides of openser simple-xmpp gateway talk at FOSDEM 2007: http://www.openser.org/events/2007-FOSDEM/slides/fosdem-2007_daniel-constan… Some details about the event: http://www.openser.org/index.php?option=com_content&task=view&id=93&Itemid=9 Cheers, Daniel

1 0

[Users] Registration of Polycom SoundPointIP phone with OpenSER
by Sheetal Bhogale 26 Feb '07

26 Feb '07

Hello all, I am trying to get this Polycom phone to register with TLS but with no luck. Can anyone help me with what certificate are you guys using,is it ur own or the defalut certificate.This phone doesn't seem to be liking my certificate netiher its able to register with the default certificate. Also anything with the firmware since I am not able to get this phone registered as well. Any help would be appreciated. Thanks, Sheetal

1 0

[Serusers] Strange Via parsing error....
by sip 26 Feb '07

26 Feb '07

I'm getting an odd Via parsing error from SER 0.9.6: Feb 23 16:52:49 death ser[17389]: error: parse_via_param Feb 23 16:52:49 death ser[17389]: ERROR: parse_via on: <sip/2.0/udp 172.30.237.149:56755;branch=z9hg4bk-d87543-d75bc86d826ac929-1--d87543-> Feb 23 16:52:49 death ser[17389]: ERROR: parse_via parse error, parsed so far:<sip/2.0/udp 172.30.237.149:56755;branch=z9hg4bk-d87543-d75bc86d826ac929-1--d87543-> Feb 23 16:52:49 death ser[17389]: ERROR: get_hdr_field: bad via Feb 23 16:52:49 death ser[17389]: ERROR: parse_msg: message=<REGISTEl sip:proxy.ideasip.com SIP/2.0> Feb 23 16:52:49 death ser[17389]: ERROR: receive_msg: parse_msg failed When I look at the packet, it looks like the actual SIP data is somehow getting garbled... with odd characters showing up in the middle of headers, etc. Any idea what might cause this? U 148.233.151.30:43764 -> XX.XX.XX.XX:5060 REGISTEl sip:proxy.ideasip.com SIP/2.0. Via: sip/2.0/udp 172.30.237.149:16240;branch=z9hg4bk-d87543-9332b73b5700e95c-1--d87543-. Max-Forward2a:70. Contactm:<sip:user@148.233.151.30:32332;rinstance=6c0c8351d99e79db>. To: "mario"<sip:user@proxy.ideasip.com>. From: "mario"<sip:user@proxy.ideasip.com>;tag=fe132761. Call-ID: n2m5owi1odc4mzkznmm5mjflmzvmzmu3zgjmngqym2y.. CSe1h:1 register. Expire1k:3600. Allo0b:invite, ack, cancel, options, bye, refer, notify, message, subscribe, info. User-Agen5m:x-lite release 1006e stamp 34025. Content-Lengthl:0. As you can see... things like REGISTEl, Max Forward2a: Expire1k: All these things look garbled. Would this be a transmission error of some kind (the garbled headers are identical for each submitted packet, though, so it seems unlikely) ? N.

4 3

[Users] failover routing and LCR
by Howard Tang 26 Feb '07

26 Feb '07

Hi, I would like to set up LCR and failover routing in my openser. I want to find some working examples on how i can set that up. Can someone point me to the right direction please? If any one can offer me a sample config that will be great. -- Howard Tang

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[Users] Modules Index Updated
by Edson 26 Feb '07

26 Feb '07

I'm glad to inform all the members of the list that ALL modules, from ALL OpenSER versions (0.9.x, 1.0.x, 1.1.x and 1.2.x) have been updated and included on the Wiki page (http://www.openser.org/dokuwiki/doku.php/start). All index are presented in a table format with three columns: the first has the name of the 'thing' being referenced; the second the module from where that reference came from; and the third is for user comments. All indexes were generated after the files on the web, so it's possible that there are some inconsistences between this files and the ones on CVS. Please let me know about any update, so that I can re-run the script to update the indexes. Due to automatization, some documentation problems pops up. like the use of a '$' sign in front of a MI command on the dialog module. They are others. what I found I corrected manually.. So, feel free to use it, and don't forget to re-inform the list about any changes on the original files. Sugestions, as always, are welcome. ;) Edson.

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Fwd: Re: [Serusers] xlite and ser
by zahra bahar 26 Feb '07

26 Feb '07

Note: forwarded message attached. ____________________________________________________________________________________ We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265

1 0

[Serusers] Problem with SER
by Jesús Terán 26 Feb '07

26 Feb '07

Hello, i am using the ser 0.8.12 and i am getting the next error in the logs: Feb 22 09:47:13 117 /usr/sbin/ser[11779]: ERROR: warning_builder: buffer size exceeded Feb 22 09:47:13 117 /usr/sbin/ser[11779]: WARNING: warning skipped -- too big Feb 22 09:47:16 117 /usr/sbin/ser[11779]: ERROR: warning_builder: buffer size exceeded Feb 22 09:47:16 117 /usr/sbin/ser[11779]: WARNING: warning skipped -- too big And when this happen some process block and dont permit to make and to do calls. Can anyone help me? My ser is on a Red Hat enterprise linux 4

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RE: [Users] cancelling INVITE during OK response
by Web Monkey 26 Feb '07

26 Feb '07

hi, thanks for your response. is the sending of CANCEL handled by the openser engine internally? i need to do this from the openser config file. basically, my statement is like this: onreply_route[2] { if (status =~ "(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); append_hf("P-hint: Reply applied\r\n"); } use_media_proxy(); } if (status == "200") { invoke_my_function_from_here("$fu:$tu:$ci"); if (retcode == 1) { # propagate OK to caller; do nothing } else { # terminate or cancel INVITE from here } } } i should also be able to do this in proxy1 from the figure below. i think i need to send CANCEL to UAC1 and BYE to UAC2. this is my understanding of the RFC. someone may have implemented a function or module to handle this. thanks again. "Premnath, KN" <Premnath.KN(a)siemens.com> wrote: Hi Sho, When you get 200OK, you send an "ACK", so it is a good indication for the initial "INVITE" that it is ACKnowledged. When you get an error, anyway you will not send "ACK" instead still send the "CANCEL" so the UAC2 is clearly handled (and it will not wait for ACK), give a try !!! Thank You,. With warm regards, Premnath Narayanan --------------------------------- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Web Monkey Sent: Monday, February 26, 2007 9:24 AM To: users(a)openser.org; users(a)iptel.org Subject: [Users] cancelling INVITE during OK response hi, is there a way to cancel or terminate an INVITE or call upon receipt of the OK reply. my configuration invokes an external program which either return success or failure for OK message (response to INVITE). when failure is returned, i want the script to terminate the call or cancel the INVITE (OK message will not be forwarded to the calling UA). is there an existing function or module to realize this? the scenario is drawn below with provisional messages removed. UAC1 ------------------ proxy1 ----------------- proxy2 --------------- UAC2 call: INVITE ---------------------------> INVITE ----------------------> INVITE -----------------> OK <---------------- *XXX - error occurred during processing OK message thanks, sho --------------------------------- It's here! Your new message! Get new email alerts with the free Yahoo! Toolbar. The new company Nokia Siemens Networks is expected to start operations in the first quarter of 2007, subject to customary regulatory approvals, the completion of standard closing conditions, and the agreement of a number of detailed implementation steps. Disclaimer: This e-mail may contain trade secrets or privileged, undisclosed or otherwise confidential information. If you have received this e-mail in error, you are hereby notified that any review, copying or distribution of it is strictly prohibited. Please inform us immediately and destroy the original transmittal. Thank you for your cooperation. --------------------------------- Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.

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[Serusers] SER Users and Developers Meeting in Prague, March, colocated with the IETF
by Jiri Kuthan 25 Feb '07

25 Feb '07

Ladies and gentlemen, I'm glad to invite you to a meeting of SER users and developers in March 2007, in Prague, Czech Republic. See more at http://www.iptel.org/ietf_meeting I'm looking forward to seeing you there. In the meantime, please register yourself if you plan to attend and don't hesitate to make any agenda suggestions -- SER related ideally, even though any "social" suggestions are perfectly fine too. I can promise you it will be a wonderful event -- Prague is beatiful, SER is beatiful, and so is the folks involved in SER development -- LET'S MEET! -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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[Users] mediaproxy exceptions on session delete
by Klaus Darilion 25 Feb '07

25 Feb '07

Hi! I have a problem with mediaproxy 1.8. Whenever a session is deleted there is an exception: Feb 14 14:18:27 foobar mediaproxy[27053]: delete OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU. info= Feb 14 14:18:27 foobar mediaproxy[27053]: session OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU.: 10558/10557/21113 packets, 778927/774750/1553531 bytes (caller/called/relayed) Feb 14 14:18:27 foobar mediaproxy[27053]: session OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU.: ended. Feb 14 14:18:27 foobar mediaproxy[27053]: error: uncaptured python exception, closing channel <rtphandler.CommandHandler connected 5.6.7.8:38879 at 0xb7d255ec> (exceptions.NameError:global name 'StopRecordSerializer' is not defined [/usr/lib/python2.3/asyncore.py|readwrite|86] [/usr/lib/python2.3/asyncore.py|handle_read_event|390] [/opt/mediaproxy/modules/rtphandler.py|handle_read|441] [/opt/mediaproxy/modules/rtphandler.py|process|568]) Feb 14 14:19:53 foobar mediaproxy[27053]: status Feb 14 14:19:53 foobar mediaproxy[27053]: execution time: 0.11 ms I could not find the problem - accounting.py with StopRecordSerializer is there, and it is imported in rtphandler. Do others also have problems and maybe a workaround? regards klaus -- Klaus Darilion nic.at

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Fwd: [Serusers] xlite and ser
by zahra bahar 25 Feb '07

25 Feb '07

Note: forwarded message attached. ____________________________________________________________________________________ Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367

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[Serusers] reminder: PRague meeting registration deadline approaching
by Jiri Kuthan 24 Feb '07

24 Feb '07

dont forget to register yourself if you plan to attend -- we really need to dimension the meeting facilities properly. http://www.iptel.org/ietf_meeting/participants Looking forward to seeing ya all! -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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[Users] new CDRTool release 5.0.10
by Adrian Georgescu 24 Feb '07

24 Feb '07

Hello, CDRTool 5.0.10 has been released. The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog from 5.0.5: cdrtool (5.0.10) unstable; urgency=low * Fixed reload of prepaid accounts from rating page to reload only the prepaid account that has changed (when PHP register_globals if off) cdrtool (5.0.9) unstable; urgency=low * Added /var/www/CDRTool/scripts/replicationStatus.php script to easily check the database replication process Add to global.inc $CDRTool['replicated_databases']=array ('DB1','DB2'); and defined DB1 and DB2 connections to the mysql databases that replicate to each other. The connections must use the same IP addresses and usernames used during the setup of the replication process cdrtool (5.0.8) unstable; urgency=low * Enable failover between CDR databases for normalization process db_class can be an array with database connection classes * Removed db_class_readonly from global.inc cdrtool (5.0.7) unstable; urgency=low * Fixed version to be compliant with debian native version numbering cdrtool (5.0-6) unstable; urgency=low * Avoid unnecessary sql OR clause in CDR search that caused slow queries Regards, Adrian Georgescu

1 0

Re: [Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 24 Feb '07

24 Feb '07

i had : UA1 (192.168.4.66:5061) }------------->Asterisk----------->SER-----------> UA3 (192.168.4.65:5061) UA2 (192.168.4.66:5062) then UA1 moved to SER domain : UA3 (192.168.4.65:5061) UA2 (192.168.4.66:5062) ------------->Asterisk----------->SER-----------> UA1 (192.168.4.66:5061) UA1 registers to SER and SER forwards the registration to ASTERISK, Asterisk is aware of the new location. UA2 tries to call UA3. It sends the invite to Asterisk and Asterisk replies with a TRYING message. after a while the call fails with an error 2007/2/24, ram <talk2ram(a)gmail.com>: > > then register with SER, should do the job > > UA----ASterisk----SER-----Phone > > is this correct > > ram > > > On 2/24/07, karim basraoui <basraouik(a)gmail.com> wrote: > > > > Hi, > > > > the call i'm trying to establish is from ASTERISK to SER. > > the UA i'm calling was initially in ASTERISK domain and moved to SER > > domain , a roaming case. > > > > 2007/2/24, ram < talk2ram(a)gmail.com>: > > > > > > Hi > > > > > > your question is not clear > > > > > > how is the call path happening from ASterisk or from SER > > > or both are gateway proxies ? > > > > > > ram > > > > > > > > > On 2/23/07, karim basraoui < basraouik(a)gmail.com > wrote: > > > > > > > Hi, > > > > > > > > I've set up a small architecture in which an ASterisk proxy is > > > > connected to a SER proxy in two different domains. > > > > > > > > I have sip phones connected to both of them. when an asterisk > > > > subscriber "travels" to SER domain it tries to register to SER. I've > > > > configured SER to accept foreign > > > > subscribers and to REGISTER them to their initial domain by the > > > > command "t_replicate". Till now everything works good, the UA is registered > > > > and Asterisk is aware > > > > of its new location, it displays in peers list the UA and its new IP > > > > address and port. > > > > > > > > The problem is when i'm trying to call the foreign UA from an > > > > Asterisk subscriber, it doesn't work. i have a TRYING response but no call > > > > establishement. > > > > > > > > how can i overcome this problem please? > > > > > > > > -- > > > > BASRAOUI Karim > > > > www.basraouik.afrikart.net > > > > _______________________________________________ > > > > Serusers mailing list > > > > Serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > -- > > BASRAOUI Karim > > www.basraouik.afrikart.net > > > > -- BASRAOUI Karim www.basraouik.afrikart.net

1 0

Re: [Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 24 Feb '07

24 Feb '07

Hi, the call i'm trying to establish is from ASTERISK to SER. the UA i'm calling was initially in ASTERISK domain and moved to SER domain , a roaming case. 2007/2/24, ram <talk2ram(a)gmail.com>: > > Hi > > your question is not clear > > how is the call path happening from ASterisk or from SER > or both are gateway proxies ? > > ram > > > On 2/23/07, karim basraoui <basraouik(a)gmail.com> wrote: > > > Hi, > > > > I've set up a small architecture in which an ASterisk proxy is connected > > to a SER proxy in two different domains. > > > > I have sip phones connected to both of them. when an asterisk subscriber > > "travels" to SER domain it tries to register to SER. I've configured SER to > > accept foreign > > subscribers and to REGISTER them to their initial domain by the command > > "t_replicate". Till now everything works good, the UA is registered and > > Asterisk is aware > > of its new location, it displays in peers list the UA and its new IP > > address and port. > > > > The problem is when i'm trying to call the foreign UA from an Asterisk > > subscriber, it doesn't work. i have a TRYING response but no call > > establishement. > > > > how can i overcome this problem please? > > > > -- > > BASRAOUI Karim > > www.basraouik.afrikart.net > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- BASRAOUI Karim www.basraouik.afrikart.net

1 0

[Serusers] iptel.org ANNOUNCEMENT: New SER bug marshal
by Greger V. Teigre 24 Feb '07

24 Feb '07

Hi all! I'm pleased to announce that Lakmal Silva has volunteered to be our SER bug marshal! Lakmal's main task will be to interface with the community and developers and make sure that bug fixes and feature requests don't disappear on the serusers and serdev mailing lists, as well as get them properly described/documented. (Please continue to use the tracker or mailing lists as you are used to.) The bug marshal's responsibilities are in more details: 1. Scan the mailings lists for bug reports, patches, feature requests, and improvement suggestions and: a) make sure they are registered on either http://iptel.org/ser/wishlist or as a tracker item (unless a developer picks it up) with unscheduled status b) request more information/debug traces/patches/etc in order to make the item possible to act upon for a developer c) solicit input from the developer(s) if clarification is needed 2. Pick up discussions on the mailing lists that concern new features, put new features on the wish list and if appropriate create a new specification/discussion page that will specify the feature in more detail and can be used for discussion purposes (examples: http://www.iptel.org/ser/specifications) Please don't contact Lakmal in private emails, but use the mailing lists if you have questions/issues. On behalf of the SER community, I welcome Lakmal and wish him luck in his new role as SER bug marshal! Greger

4 7

[Serusers] SER 0.9.6 with H.264 Problem!
by Steven Wong 24 Feb '07

24 Feb '07

Hello All: You will find enclosed our ser.cfg file that I have adapted from a very generous SIP user from the Internet. This ser.cfg file work perfectly with Audio (IP Phone and Softphone). But when I am trying to use a Video Phone (we did try with GrandStream GXV-3000 Video Phone), it just giving us Audio but NOT Video. I did also try the same GrandStream GXV-3000 Video Phone with FreeWorldDialup and it is working fine with Video. We have checked the etherreal output while trying to make a call and we are having this type of errors on the etherreal: ----- RTP Protocol Source IP (The IP of the Video Phone) and Destination IP (The IP of the SER Server) ------ Payload Type = Unknown (126) ICMP Protocol Source IP (The IP of the SER Server) and Destination IP (The IP of the Video Phone) ------Destination Unreachable (Port Unreacheable) These 2 errors repeated six time before it stop with a good signal of: Payload Type = ITU-T G.711 PCMU Then the audio start but without audio. --- Would you please tell us what is going wrong with this ser.cfg file? your answer is greatly appreciated. Many thanks, Steven Wong -------------------------------------------------------------------------- debug=3 fork=yes log_stderror=no check_via=no dns=no rev_dns=no listen=XXX.XXX.XXX.XXX # INSERT YOUR IP ADDRESS HERE # port=5060 children=3 alias="HOST.DOMAINNAME.COM" alias="XXX.XXX.XXX.XXX" fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:PASSWORD@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:PASSWORD@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) #---Accounting params modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; setflag(1); if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("HOST.DOMAINNAME.COM", "subscriber")) { www_challenge("HOST.DOMAINNAME.COM", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } -------------------------------------------------------------------------- --------------------------------- The fish are biting. Get more visitors on your site using Yahoo! Search Marketing.

3 4

[Serusers] How to start 2 SER at once?
by Meidiana Ten 23 Feb '07

23 Feb '07

Hi there, I have problem in starting 2 SER at one machine at a time Can you guys show me how to start it.. I make 1 SER for home proxy (ser.cfg) and the other for SEMS (ser_sems.cfg) on same folder then I start ser.cfg with command #ser, it works well.. but when I try to start ser_sems.cfg with command #ser_sems, it doesn't work. It said it's unknown command Please show me where do I get wrong Thanx Regards, Meidiana ____________________________________________________________________________________ Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091

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