- sr-users - kamailio.org

[Serusers] (no subject)
by KUMAR 23 Mar '07

23 Mar '07

thank you, that was helpful. kumar

1 0

[Serusers] Sip Softphone
by Jai Rangi 23 Mar '07

23 Mar '07

Hello, We are looking for a sip based softphone that can be customized according to our need. Something like Skype, so that use have to enter only username and password. If it can be open source that will be great. If not we can also pay some one who can do that for us. Please let me know if any one can do this or suggest me where to find that. Thank you, -JP

4 3

[Users] openser1.2.0+diameter
by 赵俊杰 23 Mar '07

23 Mar '07

Hello,ereryone I want to use diameter for sip accounting under ipv6.But I don't know how to configure it .Who can give me some suggestion? Thank you very much.My sip server is openser 1.2.0. Zhao Junjie 2007-03-22

2 1

[Serusers] groups
by KUMAR 23 Mar '07

23 Mar '07

yes, i was talking about the 'group' module. I still see it in 0.9.6. Moreover, there's even a table 'grp' in the ser database. Even if it has been removed now, i would like to know what it is/was used for. thank you kumar

2 1

[Serusers] Groups
by KUMAR 23 Mar '07

23 Mar '07

can you please tell me or point me to a link as to what are groups used for in SER. thank you kumar

2 1

[Serusers] INFO: Prague User Group Meetings Presentations Now Available
by Greger V. Teigre 23 Mar '07

23 Mar '07

To all of you who participated in the Prague User Group meetings on Monday and Wednesday: It was great to meet you, get a face behind the the name, and thanks for all the good discussions we had! There were people I didn't have the chance to talk to, I'm sure we get the opportunity another time :-) To those of you who did not have the possibility to attend: The presentations from Monday can be found on iptel.org in PDF format, together with a brief summary. http://www.iptel.org/ser/usergroup/prague/summary The Wednesday meeting was available as an mp3 stream and many people listened in and participated on IRC. http://www.iptel.org/ser_workshop The full workshop will soon be made available, as well as a summary of the conclusions and most important topic. A separate announcement will be made when available. g-)

2 1

[Serusers] SIP Accounting Module problem
by POON Ben 23 Mar '07

23 Mar '07

Hi folks, I'm trying to get my accounting services working, but am having trouble doing so... I am using the sample ser.cfg file, and am having a problem loading the acc.so module. In the ser.cfg file, I try added the line: loadmodule "/usr/sfw/lib/ser/modules/acc.so" to the bottom of the loadmodule section. I haven't done any modparam or scripting yet, even so, when I restart the ser service, I get: Starting SER : cat: cannot open /var/run/ser.pid. If I comment or remove the loadmodule line, ser will load again. Is there anything else I need to add or configure before loading this module? I went to /usr/sfw/lib/ser/modules to make sure acc.so is there, and it is. Any help is greatly appreciated. Thank you very much. Ben

3 7

[Users] OpenSER not sending responses to IP in via header
by Tim Madorma 23 Mar '07

23 Mar '07

Hi, I have an OpenSER behind a BIG IP load balancer. The scenario is as follows: 1) The UAC will initiate the call with an INVITE that has the REQ URI set to the IP of the BIG IP. 2) The OpenSER has an alias configured for the IP address of the BIG IP so it receives the incoming INVITE, but when it sends a 100 response, it does not send it to the UAC address that is specified in the VIA header, rather it sends it to the source IP (which is the IP of the BIG IP load balancer). This is undesirable. 3) After the UAS receives the proxied INVITE, it responds with a 100, then answers with a 200. 4) When the OpenSER receives the 100, it forwards this back to the BIG IP load balancer! First, I am surprised that it fowards a 100 at all, and second it is undesirable to send any responses to the BIG IP load balancer since the via header indicates the IP of the UAC. 5) When the OpenSER receives the 200, it forwards it back to the BIG IP also.. again, I don't know why it isn't sending to the IP address of the via - which is that of the UAC. Any suggestions would be most helpful. Here are my related openser.cfg settings: # LOCAL IP ADDRESS AND PORTS listen=65.185.232.62:5061 listen=65.185.232.62:5062 listen=65.185.232.62:5063 # IP ADDRESS AND PORTS OF THE BIG IP LOAD BALANCER alias=65.185.233.104:5061 alias=65.185.233.104:5062 alias=65.185.233.104:5063 thanks, Tim

3 3

[Serusers] group info
by KUMAR 23 Mar '07

23 Mar '07

Hi, I want to store the buddylist from the client application on the SER. How do i go about it. I guess, i can do it using the groups but cant' really find any resources to do the studying part. Can you please point me as to where i can find the information regarding this.?? thank you kumar

2 1

[Users] OpenSER sometimes forwarding 100 response
by Tim Madorma 23 Mar '07

23 Mar '07

Hi, I'm using version 1.2 of OpenSER. After proxying an INVITE from a UAC to a UAS, the UAS replies with a 100 Trying. *Sometimes* the OpenSER forwards this 100 Trying to the UAC. Any idea how to disable this? I think this can be done using onreply_route, but it seems that a received 100 should not be forwarded by default. thanks, Tim

2 1

[Users] reason of interlocking with Asterisk server
by 김형선 23 Mar '07

23 Mar '07

Hello. I think that some sites use OpenSER with Asterisk server. But Why? (for accounting? or supplementary service?) Aren't they same? In conceptionally, OpenSER is central VoIP server, and Asterisk is IP-PBX for apart site. right? What is the structure of service being applied, in generally? OpenSER, Asterisk,... What is the role of them? thank you. 새로운 기부문화의 씨앗, 해피빈

2 1

[Users] Openser + postgres not starting
by Knight Tiger 23 Mar '07

23 Mar '07

Hi I have installed Openser v 1.2 no tls. I have Postgres version 8.0.7. I tried openser start [root@openser /usr/local/lib]# openser start ERROR: bad config file (1 errors) [root@openser /usr/local/lib]# openser -c 0(29611) 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(29611) parse error (41,13-14): failed to load module ERROR: bad config file (1 errors) 0(29611) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy I then ran openserctl moni cycle #1, if constant make sure server lives My openser.cfg file # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "postgres.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ...... .... I checked for the PQescapeStringConn and found this www.mail-archive.com/devel@openser.org.msg05182.html but I could not make anything out of it I dont want authentication just yet. Can anyone tell me what is wrong ? Thanks in advance Caugar

3 3

[Users] an example of xcap_table entry?
by jh＠tutpro.com 22 Mar '07

22 Mar '07

is there somewhere an example off xcap_table entry that would show who is allowed to subscribe the presence of an AoR? -- juha

3 2

[Users] OpenSER performance tests
by Bogdan-Andrei Iancu 22 Mar '07

22 Mar '07

Hi everybody, based on the newly 1.2.0 OpenSER release, we ran some tests in order to give some realistic numbers about the performances of OpenSER. I think it is a very useful information for our community or other interested people. For more, please refer to the news bulletin: http://www.openser.org/mos/view/News/NewsItem/OpenSER-1.2.0---Perfomance-Te… or directly to the tests results web page: http://www.openser.org/docs/openser-performance-tests/ regards, bogdan

6 11

[Serusers] ACK for non-2XX not record-routed
by Emmanuel Hislen 22 Mar '07

22 Mar '07

Hi, I've been using SER (0.9.6) as a Proxy doing Loose Routing between 2 endpoints. If I place a call I can see how the ACK for the 2XX response to the INVITE is properly handled (Route header removed, Record-Route inserted). Now with the call still up, I do a re-INVITE which is rejected with a 415. The re-INVITE request sent by UAC contained a Route header, as it was an in-dialog request in a record-routed dialog. Now as per RFC3261 section 17.1.1.3: If the INVITE request whose response is being acknowledged had Route header fields, those header fields MUST appear in the ACK. This is to ensure that the ACK can be routed properly through any downstream stateless proxies. My UAC complied with the above, but SER seemed to choke on this: whereas the ACK for 2XX was successfully routed by SER in the original INVITE, the ACK for non-2XX (looking exactly the same: same Request URI, same IP destination, same Route header, ... different Via branch of course) is not handled properly: yes the ACK reaches UAS but it still has the Route header in it, and no Record-Route was added. And I'm guessing that if there had been another proxy in the Route Set on the way to the UAS it would have been bypassed. SER traces for ACK-2XX: > 0(13574) ===> Loose Routing logic... > 0(13574) parse_headers: flags=-1 > 0(13574) DEBUG: t_newtran: msg id=89 , global msg id=88 , T on entrance=0xffffffff > 0(13574) parse_headers: flags=-1 > 0(13574) parse_headers: flags=60 > 0(13574) t_lookup_request: start searching: hash=41794, isACK=1 > 0(13574) parse_headers: flags=28 > 0(13574) DEBUG: t_lookup_request: e2e proxy ACK found > 0(13574) parse_headers: flags=4 > 0(13574) DEBUG: totag for e2e ACK found: 0 > 0(13574) SER: forwarding ACK statelessly SER traces for ACK-non-2XX: > 0(13574) ===> Loose Routing logic... > 0(13574) parse_headers: flags=-1 > 0(13574) DEBUG: t_newtran: msg id=92 , global msg id=91 , T on entrance=0xffffffff > 0(13574) parse_headers: flags=-1 > 0(13574) parse_headers: flags=60 > 0(13574) t_lookup_request: start searching: hash=41807, isACK=1 > 0(13574) DEBUG: RFC3261 transaction matched, tid=007d7195499e33a1 > 0(13574) DEBUG: t_lookup_request: transaction found (T=0xb616ae48) > 0(13574) DEBUG: cleanup_uac_timers: RETR/FR timers reset > 0(13574) DEBUG: add_to_tail_of_timer[2]: 0xb616ae90 > 0(13574) DEBUG:destroy_avp_list: destroying list (nil) > 0(13574) receive_msg: cleaning up This doesn't seem right, am I missing something here? Thanks, Emmanuel.

3 5

Re: [Users] To: header in e.164 format - adding a plus
by Daryl Sanders 22 Mar '07

22 Mar '07

I'd love to use uac_replace_to(), but to my knowledge it does not exist. I read through the UAC module docs and there is no mention of it. Any other posibilities? If I were to use subst(), can it be applied it in such a way that it does not cause problems with the dialog? - Daryl On 3/21/07, Alan Crosswell <alan(a)columbia.edu> wrote: > Let me guess -- Level 3? They're the only ones that make that bogus > requirement AFAIK. And, the subst() is of course wrong, you need to do > uac_replace_to() or break the dialog otherwise.... > /a > > Daryl Sanders wrote: > > One of my providers requires the To: header to be in e.164 format with > > a "+" in front of it. Since there is no uac_replace_to() function yet, > > what are my options for adding this plus symbol? > > > > Before anyone suggests it... I cannot just use another provider. I > > must find a solution to add the plus symbol. > > > > Thanks, > > Daryl > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users >

3 3

[Users] SUBSCRIBE/NOTIFY problem
by Vadim Berezniker 22 Mar '07

22 Mar '07

The setup is: Polycom 601 => OpenSER => Asterisk The SUBSCRIBE from the phone reaches asterisk just fine and the OK is returned properly to the phone. The NOTIFY messages don't get to the phone, however. SER sends it to the requested URI (which is phone@ser), which ends up being relayed back to Asterisk.

1 1

[Users] parameter of route(n)
by jh＠tutpro.com 22 Mar '07

22 Mar '07

is it so that currently parameter n of call route(n) needs to be an integer constant, i.e., that use of a pseudo variable is not possible? -- juha

4 8

[Users] dispatcher, failure route and force_send_socket problem
by Klaus Darilion 22 Mar '07

22 Mar '07

Hi! Once again I have a problem with force_send_socket and branches in failure route. Scenario: normal route block --> call SIP client after timeout --> failure route in failure route send to gateway via dispatcher module The gateway accepts signaling only from a certain port, thus I have to use force_send_socket. The problem is, that if I call force_send_socket() in failure route, it sets the send_Socket for branches[0]. Thus if I call force_Send_socket before ds_select_domain it sets branches[0], but the sendsocket is not copied into the new branch when ds_select_domain internally calls append_branch. With version 1.1 I can use the ugly workaround to manually call append_branch too, as this copies another branch and also copies the sendsocket. Then I have 2 new branches where branches[1] gets dropped by teh firewall (wrong sendseocket) but branches[2] succeeds. But with version 1.2 the sendsocket is never copied from branches[0] and I can't set it manually for a certain branch. Are there any workarounds (for version 1.1. and 1.2) thanks klaus version 1.1: -- Klaus Darilion nic.at

1 1

[Users] [user] openser TLS and eyebeam
by Bodin Bruno 22 Mar '07

22 Mar '07

I made certificat, I add user folder to openser ser config (with lines it must) and I add certificat in Internet explorer for client. But eyebeam say certificat failed. Anyone have experience with eyebeam and tls ? thank.

2 1

[Users] OpenSer Asterisk
by Michael Chan 22 Mar '07

22 Mar '07

Hi, OpenSer Support, I read the OpenSer and Asterisk real time intergation and follow the config instruction but still no result that we can make PSTN call through Asterisk without using the name and PW of OpenSer on the UA Sip Client (X-lite). On the Asterisk side, we assigned a user name and PW to OpenSer, but not sure where we can create script on OpenSer side that allow all calls to Asterisk without using the shared secret name and PW of OpenSer. Please any help will be grateful. MC -------------------------------------------------------------------------------- I am using the free version of SPAMfighter for private users. It has removed 234 spam emails to date. Paying users do not have this message in their emails. Try SPAMfighter for free now!

2 1

RE: [Serusers] re: composing SIP messages
by Goh, Tee-Yong 22 Mar '07

22 Mar '07

Hi Greger, Thank you for sharing the information. May i know which part of the program in the SER can i find FIFO? Please advice. Thanks. Regards, Howard -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Greger V. Teigre Sent: Thursday, March 22, 2007 3:10 PM To: Jignesh Gandhi Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] re: composing SIP messages No, but there is a FIFO interface that allows inserting messages into SER. For anything but trivial cases, you will probably want to do it from SEMS. g-) Jignesh Gandhi wrote: Is there a way to compose SIP messages such as INVITE from scratch using SER ? I am new to SER... So any help would be appreciated thanks, -- Jignesh Gandhi jigpgandhi(a)gmail.com _____ _______________________________________________ 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

1 0

[Serusers] re: composing SIP messages
by Jignesh Gandhi 22 Mar '07

22 Mar '07

Is there a way to compose SIP messages such as INVITE from scratch using SER ? I am new to SER... So any help would be appreciated thanks, -- Jignesh Gandhi jigpgandhi(a)gmail.com

2 1

[Users] Compiling Openser for best performance on Solaris SPARC
by Sergio Gutierrez 21 Mar '07

21 Mar '07

Hi. Currently I am installing Openser 1.2.0 on a Sun Fire T2000 running Solaris 10. Although I used an optimized version of GCC for SPARC Systems, and used the recommended settings to generate accurate code for my Processor (UltraSPARC T1), the results of the performance tests are just 50% of the presented ones on the web page. My setup is the following: Proxy and UAS: Sun Fire T2000, 8 GB RAM, 1 UltraSPARC T1 1.2 GHZ 6 Core. UAC's: Two domain of a Sun Fire 12K, 4 GB RAM, 4 UlraSPARC III+ 900 Mhz Software versions listed as on the webpage. openser -V output: version: openser 1.2.0-notls (sparc/solaris) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT-NOSMP ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, select, /dev/poll. svnrevision: unknown @(#) $Id: main.c 1827 2007-03-12 15:22:53Z bogdan_iancu $ main.c compiled on 21:43:51 Mar 19 2007 with gcc 4.0.3 I would thank any idea about how could I improve the performnace of my setup. I think that because of the hardware specifications, the result should be at least equal than listed. Thanks in advance. Kind regards. Sergio Gutiérrez.

2 7

[Serusers] Acc not writing to MySQL
by POON Ben 21 Mar '07

21 Mar '07

Hi all, I am having a lot of trouble making the acc module work... I've googled and tried a lot of sample config but am still having no luck. Can someone please help me to get accounting on my server? Below is my ser.cfg file. It's basically the sample one with some editing. The auth stuff work really nicely :) Everything seems to load fine when I do a "ser -D -E" However, I noticed that nothing is ever entered into the acc table in the ser database. I am running ser 0.9.6 with mysql 5.0.37. Thank you so much! Ben # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=no # (cmd line: -E) listen=1.1.1.1 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias="localhost" alias=1.1.1.1 /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://admin:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("localhost", "subscriber")) { www_challenge("localhost", "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; }; }; # Report missed calls setflag(2); append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # Set accounting flag setflag(1); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

1 0

[Users] To: header in e.164 format - adding a plus
by Daryl Sanders 21 Mar '07

21 Mar '07

One of my providers requires the To: header to be in e.164 format with a "+" in front of it. Since there is no uac_replace_to() function yet, what are my options for adding this plus symbol? Before anyone suggests it... I cannot just use another provider. I must find a solution to add the plus symbol. Thanks, Daryl

3 2

[Users] Fresh OpenSER 1.2.0 Openserctl can only add 1 user?!
by Geir O. Jensen 21 Mar '07

21 Mar '07

I have two server, one in production and one for testing. Trying to debug something else entirely I upgraded the production server from 1.1.0 to 1.2.0 yesterday. Everything seems to work fine on it. So I wiped my testbed and compiled OpenSER 1.2.0 on the testing system, installed the mysql db and started it up. Worked fine. I then began adding testaccounts on the testsystem, - first user ok... - second user, not so ok... # openserctl add test test gojensen(a)uninett.no MySql password for user 'xxx@localhost': ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 ERROR: introducing the new user 'test' to the database failed Of course, I couldn't add a third or fourth either, so I'm basically stuck with one user and the admin account. Any idea where to start debugging this?! (As a note, I added 3 test accounts on the production server with no problems... Their config files are basically identical - the testing server only adds the presence module...) // sip:gojensen@uninett.no | h323: 004710012 pstn: +47 73 55 79 23 | fax: +47 73 55 79 01

2 2

[Users] Wesip C2C
by Helmut Kuper 21 Mar '07

21 Mar '07

Hello, after upgrading to wesip 0.1.3 I'm not able anymore to start c2c application. in WeSIP's logfile I can find this exception: 2007-03-21 18:11:52 StandardManager[/click2call-08_11_06]: Seeding random number generator class java.security.SecureRandom 2007-03-21 18:11:52 StandardManager[/click2call-08_11_06]: Seeding of random number generator has been completed 2007-03-21 18:11:52 HTMLManager: ManagerServlet.install[/click2call-08_11_06] java.lang.NoClassDefFoundError: com/voztele/wesip/apps/click2call/C2CDefaultServlet at java.lang.ClassLoader.defineClass1(Native Method) at java.lang.ClassLoader.defineClass(Unknown Source) at java.security.SecureClassLoader.defineClass(Unknown Source) at java.net.URLClassLoader.defineClass(Unknown Source) at java.net.URLClassLoader.access$100(Unknown Source) at java.net.URLClassLoader$1.run(Unknown Source) at java.security.AccessController.doPrivileged(Native Method) at java.net.URLClassLoader.findClass(Unknown Source) at java.lang.ClassLoader.loadClass(Unknown Source) at sun.misc.Launcher$AppClassLoader.loadClass(Unknown Source) at java.lang.ClassLoader.loadClass(Unknown Source) at org.apache.catalina.loader.WebappClassLoader.loadClass(WebappClassLoader.java:1379) at org.apache.catalina.loader.WebappClassLoader.loadClass(WebappClassLoader.java:1307) at com.voztele.sipservlet.core.StandardSipWrapper.loadServlet(StandardSipWrapper.java:753) at com.voztele.sipservlet.core.StandardSipWrapper.load(StandardSipWrapper.java:701) at org.apache.catalina.core.StandardHttpContext.loadOnStartup(StandardHttpContext.java:3382) at org.apache.catalina.core.StandardHttpContext.start(StandardHttpContext.java:3605) at com.voztele.sipservlet.core.ConvergedContext.start(ConvergedContext.java:259) at org.apache.catalina.core.ContainerBase.addChildInternal(ContainerBase.java:773) at org.apache.catalina.core.ContainerBase.addChild(ContainerBase.java:759) at org.apache.catalina.core.StandardHttpHost.addChild(StandardHttpHost.java:552) at com.voztele.sipservlet.core.ConvergedHost.addChild(ConvergedHost.java:157) at com.voztele.sipservlet.core.ConvergedHostDeployer.install(ConvergedHostDeployer.java:264) at com.voztele.sipservlet.core.ConvergedHost.install(ConvergedHost.java:170) at org.apache.catalina.servlets.ManagerServlet.install(ManagerServlet.java:663) at org.apache.catalina.servlets.HTMLManagerServlet.install(HTMLManagerServlet.java:265) at org.apache.catalina.servlets.HTMLManagerServlet.doPost(HTMLManagerServlet.java:245) at javax.servlet.http.HttpServlet.service(HttpServlet.java:760) at javax.servlet.http.HttpServlet.service(HttpServlet.java:853) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:200) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:146) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:206) #################################################################################### When I cwitch back to wesip 0.1.2 C2C is loading and starting fine. I use click2call-08_11_06 When I search for the missed class C2CDefaultServlet I get this: hk@vm-en-test0:~/wesip/wesip$ find . -name "C2CDefaultServlet*" ./wesipapps/click2call-08_11_06/WEB-INF/classes/com/voztele/wesip/apps/click2call/C2CDefaultServlet.class So it#s there, but wesip doesn#t find it. Wher do I have to tell weSIP the location of that class ? regards helmut

1 0

Re: [Users] NAT Problem with OpenSER behind load balancer
by Daniel-Constantin Mierla 21 Mar '07

21 Mar '07

You should keep cc-ing mailing lists, so people coming after a while, with same issue, will be reading mail thread and see that the given solution was right. Thanks, Daniel On 03/20/07 21:42, Tim Madorma wrote: > Got it. thanks so much! > > Tim > > On 3/20/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello, >> >> On 03/20/07 17:39, Tim Madorma wrote: >> > Hi, >> > >> > I have put the OpenSER version 1.2 behind a BIG IP load balancer which >> > provides a static private IP address (10.1.1.20) via NAT to the >> > OpenSER. In our application, the clients will send to one of the three >> > public addresses that are defined as aliases: >> > >> > alias="65.185.232.62:5061" >> > alias="65.185.232.62:5062" >> > alias="65.185.232.62:5063" >> you have to remove quotes around the value of alias parameters. As it is >> now, the value between quotes is considered to be hostname. >> alias=65.185.232.62:5061 >> alias=65.185.232.62:5062 >> alias=65.185.232.62:5063 >> >> Same case as it was reported by Juha, today with loose routing ... >> >> Cheers, >> Daniel >> >> > >> > An incoming OPTIONS message is received with the host in the REQ URI >> > set to 65.185.232.62:5063 is received and when it hits this section of >> > openser.cfg: >> > >> > if (!uri==myself) { >> > # mark routing logic in request >> > append_hf("P-hint: outbound\r\n"); >> > route(1); >> > }; >> > >> > it executes within the if conditional and tries to route the OPTIONS >> > message. I am wondering why uri does not match myself. The debug >> > output from the log is as follows (it seems that the alias assignments >> > are not working): >> > >> > >> > Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] >> > grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 >> > 0.1.1.20] >> > Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] >> > grep_sock_info - checking if port 5061 matches port 5063 >> > Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] >> > grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 >> > 0.1.1.20] >> > Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] >> > grep_sock_info - checking if port 5062 matches port 5063 >> > Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] >> > grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 >> > 0.1.1.20] >> > Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] >> > grep_sock_info - checking if port 5063 matches port 5063 >> > Mar 19 16:41:11 homer openser[2267]: [ID 140248 local0.debug] >> > DEBUG:check_self: host != me >> > >> > >> > Any suggestions? I have not tried this in any other version. Is this a >> > possible bug in 1.2? >> > >> > thanks, >> > Tim >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >

3 3

[Users] ENC: Presence Module - Eyebeam 1.5
by Hildebrando Trannin 21 Mar '07

21 Mar '07

Hi all, I configured the presence module in OpenSER v1.2 and it looks like ok. But when tested with EyeBeam 1.5 (buil 30037) it doesn't work. I used the WireShark to see the SIP messages and observed that in NOTIFY messages there aren't PIDF document, neither message body. Furthermore, OpenSER didn't parse xml BODY from PRESENTITY table when try to build notify. The PIDF document that EyeBeam creates and the error that OpenSER gave me when try to parser this document are below. Can someone help me with this problem? Is it a bug from OpenSER? Regards, Hildebrando PIDF Document <?xml version="1.0" encoding="UTF-8" standalone="no" ?> <pr:presence xmlns:pr="urn:ietf:params:xml:ns:pidf" entity="sip:hildebrando@voip.maritaca" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:cipid="urn:ietf:params:xml:ns:pidf:cipid" xmlns:counterpath="www.counterpath.com/presence/ext" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"> <pr:tuple id="se43ac217"> <pr:status> <pr:basic>open</pr:basic> </pr:status> <pr:note>Busy</pr:note> <rpid:user-input last-input="2007-03-19T19:21:49Z">active</rpid:user-input> <pr:timestamp>2007-03-19T19:21:49Z</pr:timestamp> </pr:tuple> <dm:person id="p0d4e4429"> <rpid:activities> <rpid:busy/> </rpid:activities> <dm:note>Busy</dm:note> </dm:person> </pr:presence> OpenSER Error 2(6498) Chegou um PUBLISH de (hildebrando) para (sip:hildebrando@voip.maritaca) 2(6498) PRESENCE:update_presentity: querying presentity 2(6498) PRESENCE:get_subs_dialog:querying database table = active_watchers 2(6498) PRESENCE:get_p_notify_body: querying presentity 2(6498) PRESENCE:agregate_xmls: ERROR while parsing xml body message 2(6498) PRESENCE:notify:dialog informations: 2(6498) [p_user]= hildebrando [p_domain]= voip.maritaca [w_user]= gustavo [w_domain]= voip.maritaca 2(6498) [event]= presence [staus]= active [expires]= 3340 2(6498) [to_tag]= 10.6503.1174328485.15 [from_tag]= 80746145

3 5

[Users] xmpp module
by Sebastian Murray-Roberts 21 Mar '07

21 Mar '07

Hi, I just have a question about the new xmpp module. I have set up an OpenSER box as a presence server using the presence module, and everything is working fine, messaging and presence. Now I would like to allow those users to communicate with users logged into a jabber server. My first question is can presence information be exchanged between OpenSER and the jabber server? If so, how is this achieved since the xmpp module only has an xmpp_send_message() method. Also, just to make sure I understand how the module works, would this be a correct summary of the module's functionality for a SIP to Jabber message with a response: User A registers with OpenSER. User B registers with jabber server (OpenFire) User A sends a message to sip:User B*<jabber_domain>@<gateway_domain> with from details : sip:User A*<openser_domain>@<gateway_domain> OpenSER forwards the request to the gateway Gateway translates the message and sends it to the jabber server Jabber server delivers message to User B Jabber server sends response to gateway with to details: User A*<openser_domain>@<gateway_domain> Gateway forwards response to sip:User A@<openser_domain> Is that more or less correct, or am I off the mark? Finally, if that is the case, would User A need to be registered on the jabber server already? I've never worked with presence before, so I'm still trying to get to grips with it. Any comments/insights are much appreciated. Many thanks, Sebastian -- Sebastian Murray-Roberts (B.Sc Hons) Server Developer Yeigo Communications http://www.yeigo.com Tel: +27 21 409 7807 Fax: +27 21 409 7050 Mobile: +27 72 183 5998 CONFIDENTIALITY NOTICE: E-MAIL This e-mail is confidential. If you receive it by mistake, please delete it and notify the sender immediately. If you are not the intended recipient, you may not use, disclose, distribute, copy or publish any information contained herein.

2 1

[Users] Commercial SIP Server
by Michel Bensoussan 21 Mar '07

21 Mar '07

Hello Is there a way to know which commercial SIP servers are based on openser or ser? Or if a specific SIP server is based on openser or ser? Thanks. Regards, Michel.

3 5

答复: [Users] ERROR 1062 (23000): Duplicate entry
by 郭军 21 Mar '07

21 Mar '07

Why don't you use openserctl add ....to add new user? -----邮件原件----- 发件人: alper ozbilen [mailto:alperbilen@hotmail.com] 发送时间: 2007年3月21日 16:45 收件人: users(a)openser.org 主题: [Users] ERROR 1062 (23000): Duplicate entry Dear All, I can not insert a new subscriber from mysql console. When I look at subscireber table, ha1 & ha1b & and phplib_id fields looks null. So, i can only insert one new subscriber. When i attempt to insert second one, i get error written below: ERROR 1062 (23000): Duplicate entry '' for key 3 Note : My insert sentence like that: insert into subscriber (username, domain, password, email_address) values ("testuser","myfon.com","testpasswd","emailoftestuser(a)myfon.com"); I am confused about what i should to do to fix null statement of ha1 & ha2 & and phplib_id fields. on the other side, there is no probem to add new user through openserctl What was the problem? I will wait your kind suggestion. Regards, Alper Ozbilen _________________________________________________________________ Exercise your brain! Try Flexicon. http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglinema… _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

7 9

[Users] RHEL/Fedora equivalents of deb packages in RPM SPEC
by Alan Crosswell 21 Mar '07

21 Mar '07

The BuildPrereq lines in the RPM SPEC file for Fedora Core (openser.spec.fedora) supplied with 1.2.0 references debian names for the package (e.g. libxml2-dev -> libxml2-devel). Before I waste too much time finding the equivalent RHEL and/or Fedora Core packages that do the job, I thought I would ask her if anyone else has done so. If not, I'll make sure to submit patches for the SPECs once I figure them out. I am really looking to build on RHEL4 but clearly will need to grab some RPMs from Fedora Core.... Please no religious rants about Debian vs. RHEL vs. Fedora vs. SuSE and so on:-) /a

3 2

[Users] info-color of openserctl
by Andreas Granig 21 Mar '07

21 Mar '07

Hi, Could we please change the color of minfo from yellow to, say, orange in openserctl? It's impossible to read on a console with white background. Cheers, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

2 1

[Users] domain module in 1.2
by Andreas Granig 21 Mar '07

21 Mar '07

Hi, In openser 1.1, when I use db_mode=0 (no caching) for domain module, I get this error: /usr/local/openser-1.1# ./sbin/openserctl domain show 500 command 'domain_dump' not available In openser 1.2, the first time I execute the command, I get nothing, the second time openserctl just hangs until I terminate it with ctrl-c: /usr/local/openser-1.2# ./sbin/openserctl domain show /usr/local/openser-1.2# ./sbin/openserctl domain show ./sbin/openserctl: line 83: /tmp/openser_fifo: Interrupted system call Terminated Same thing for "domain reload" btw. Cheers, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

4 7

[Users] Only db_mode=0 works with dbtext loaction table in 1.2
by Daryl Sanders 21 Mar '07

21 Mar '07

When using db_mode=1,2 or 3 I get the following error when starting OpenSER 1.2 using dbtext as the usrloc DB. 1(23052) preload_udomain(): Error while doing db_query 1(23052) ERROR:ul:child_init(1): Error while preloading domain 'location' 1(23052) ERROR:init_mod_child: Error while initializing module usrloc, rank 1 I am wondering if their may be a problem with the columns in the location table file. I tried two different methods of creating the initial table files... 1.) sc.dbtext dbtext /usr/local/etc/openser/dbtext/ 2.) textdb.sh create /usr/local/etc/openser/dbtext/ The first method has problems with the version file that I tried to repair by trying to mimic the mysql table structure. The second method seems to work ok with the exception of the error I mentioned above when starting OpenSER. Any help would be greatly appreciated! Thanks, Daryl

4 10

RE: [Users] DB migration
by Leonid Fainshtein 21 Mar '07

21 Mar '07

Thank you, Brad. I missed MySQL RENAME DATABASE command. It is because the option appears in MySQL starting from version 5.1 only and I have version 5.0. It seems that for older MySQL the following sequence will work: openser_mysqldb.sh migrate openser openser12 mysqldump -u root -pxxxxx -c -t --ignore-table=openser12.version openser12 > /tmp/openser12.bak openser_mysqldb.sh reinit echo "delete from subscriber;" | mysql -u openser -pxxxxx openser echo "delete from admin_priveleges;" | mysql -u openser -pxxxxx openser mysql -u openser -pxxxxx openser < /tmp/openser12.bak Leonid -----Original Message----- From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] Sent: Tuesday, March 20, 2007 7:23 PM To: Leonid Fainshtein; users(a)openser.org Subject: RE: [Users] DB migration In MySQL at least: rename database <old name> <new name> Once you've confirmed that the migration was successful, you can run a backup of the old datbase, drop it, and then rename the new one. - Brad > -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Leonid Fainshtein > Sent: Tuesday, March 20, 2007 12:43 PM > To: users(a)openser.org > Subject: [Users] DB migration > > Hi, > I didn't understand the proposed process of MySQL DB migration from > ver.1.1 to 1.2 > For example, the result of "openser_mysqldb.sh migrate > openser openser12" will be a new database openser12. > But how can I copy/rename openser12 to openser? Does it mean > that from now, all of my auxiliary software must work with openser12? > > Thank you, > Leonid Fainshtein > > > ******************************************** > This outgoing mail message was scanned by McAfee GroupShield Engine > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

2 1

[Users] Problem with dialog module
by Leonid Fainshtein 21 Mar '07

21 Mar '07

I just to tested the issue discussed in http://openser.org/pipermail/users/2007-March/009777.html (OpenSer removes the did parameter from re-INVITE) with version 1.2.0. Unfortunatelly the problem still exists... Leonid ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

1 0

[Users] Presence and Postgresql
by H Quintana 21 Mar '07

21 Mar '07

Hi everybody, I'm using Xlite 3.0 and postgresql 8.2.3, the information about the presentity is properly stored in the table. The query brings the value for the field "body" but after the "conversion" to string the value is empty. Is Someone running the presence module with postgres? The log is at: http://cut.and.paste.org/index.php?id=952 I added this to notify.c in the line 514: DBG("PRESENCE:NOTIFY:Body array n=%d, content:<%.*s>\n", i, body_array[i]->len, body_array[i]->s); and in the log you will find: 0(14253) PRESENCE:NOTIFY:Body array n=0, content:<> before the line: 0(14253) PRESENCE:agregate_xmls: ERROR while parsing xml body message Thanks for your help, Humberto Quintana ____________________________________________________________________________________ TV dinner still cooling? Check out "Tonight's Picks" on Yahoo! TV. http://tv.yahoo.com/

3 3

[Users] lcr, branch route and setdsturi
by Andreas Granig 21 Mar '07

21 Mar '07

Hi, Is it possible to set the dst-uri in a branch-route (in openser 1.1)? It would be useful if you could for example do least-cost-routing with lcr, where some of the gateways need an extra hop via an outbound-proxy, which could be marked with a prefix in the gw-table, like: route[1] { # let's assume uri = 'sip:12345@something' load_gws(); next_gw(); # lcr module selects a gw according to lcr patterns, and some of # them needs an outbound-proxy, which are marked by prefix 'out' t_on_failure("1"); t_on_branch("1"); # do something else and t_relay() } failure_route[1] { if( /* error code which triggers failover */ ) { if(next_gw()) { t_on_branch("1"); t_on_failure("1"); t_relay(); } else { /* error */ } } } branch_route[1] { # check if it's marked for using an outbound proxy: if(uri =~ "^sip:out[0-9]+@") setdsturi("sip:my.outbound.ip:5060"); } Would this work? Comments? Regards, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

1 1

[Serusers] NAT Handling - SER behind a router
by TZieleniewski 21 Mar '07

21 Mar '07

Hi!! Right now I have SER based voip system but inside the local network. I would like to enable the external internet connection. Is it possible to realize the NAT handling but with SER-RTPProxy or SER-MediaProxy not with the public IP so still in the local net but with the strict forwarding rules on the router. where router ofcaurse has the public IP So that for instance having SER and media proxy inside local network but with whole SIP traffic routed to SER and RTP to media proxy?? Is one of the above methods more recommened at the moment with ser2?? Thank you in advance Best Tomasz

3 3

[Serusers] Call forwarding
by Jai Rangi 21 Mar '07

21 Mar '07

Hello, I have a ser and asterisk setup. Users register on ser and all the national and international calls go through asterisk. Asterisk handles the billing part for long distance calls. Now I got an issue with call forwarding. Here is the situation. Useragent A calls Usergent B. User B has setup the call the call forwarding to his cell phone. So when user A calls user B, ser check that user B has set the forwarding and send the call to Asterisk box. Asterisk box sees user A in the callerID and bills user A for that call. We want user B to be billed for that call, cause he is forwarding the calls to his cell phone (long distance number). How can I achieve this. Thank you, -Jai

5 5

Re: 答复: [Users] ERROR 1062 (23000): Duplicateentry
by sukerry 21 Mar '07

21 Mar '07

郭军,您好！ please modify openserctrlrc, set ?WEB?(some what) to be yes ======= 2007-03-21 16:49:35 您在来信中写道：======= >Why don't you use openserctl add ....to add new user? > >-----邮件原件----- >发件人: alper ozbilen [mailto:alperbilen@hotmail.com] >发送时间: 2007年3月21日 16:45 >收件人: users(a)openser.org >主题: [Users] ERROR 1062 (23000): Duplicate entry > >Dear All, > >I can not insert a new subscriber from mysql console. >When I look at subscireber table, ha1 & ha1b & and phplib_id fields looks >null. So, i can only insert one new subscriber. When i attempt to insert >second one, i get error written below: > >ERROR 1062 (23000): Duplicate entry '' for key 3 > >Note : My insert sentence like that: >insert into subscriber (username, domain, password, email_address) values >("testuser","myfon.com","testpasswd","emailoftestuser(a)myfon.com"); > >I am confused about what i should to do to fix null statement of ha1 & ha2 & >and phplib_id fields. > >on the other side, there is no probem to add new user through openserctl > >What was the problem? I will wait your kind suggestion. > >Regards, >Alper Ozbilen > >_________________________________________________________________ >Exercise your brain! Try Flexicon. >http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglinema… > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　sukerry 　　　　　　　　sukerry(a)126.com 　　　　　　　　　　2007-03-21

1 0

[Users] ERROR 1062 (23000): Duplicate entry
by alper ozbilen 21 Mar '07

21 Mar '07

Dear All, I can not insert a new subscriber from mysql console. When I look at subscireber table, ha1 & ha1b & and phplib_id fields looks null. So, i can only insert one new subscriber. When i attempt to insert second one, i get error written below: ERROR 1062 (23000): Duplicate entry '' for key 3 Note : My insert sentence like that: insert into subscriber (username, domain, password, email_address) values ("testuser","myfon.com","testpasswd","emailoftestuser(a)myfon.com"); I am confused about what i should to do to fix null statement of ha1 & ha2 & and phplib_id fields. on the other side, there is no probem to add new user through openserctl What was the problem? I will wait your kind suggestion. Regards, Alper Ozbilen _________________________________________________________________ Exercise your brain! Try Flexicon. http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglinema…

1 0

[Users] OpenSER and FWD
by Michel Bensoussan 21 Mar '07

21 Mar '07

Hello. How can I configure OpenSER to outgoing all call to FWD and more important to ensure that all INCOMING call will go through OpenSER? Thanks. Regards, Michel.

4 8

[Users] RPID single quotes problem
by Daryl Sanders 20 Mar '07

20 Mar '07

Is there an easy way to strip the single quotes from RPID in DBTEXT? When I use sc.dbtext it automatically adds single quotes to the entry like so... '17145551212' As a result when I add RPID to my header with the folowing line, I get the single quotes in the header. append_hf("P-Asserted-Identity: \"User\" <sip:+$avp(s:rpid)@x.x.x.x>\r\n"); Regards, Daryl

1 1

Re: [Users] loose_route problem with 1.2
by jh＠tutpro.com 20 Mar '07

20 Mar '07

it turned out that if i have in openser.cfg listen=x.x.x.x alias="sip.foo.bar:5060" port=5060 auto_aliases=no then openser advertises at startup: Aliases: *: sip.foo.bar:5060:* and loose_route DOES NOT recognize that Route header <sip:sip.foo.bar;lr;transport=TCP> is local. on the other hand, if i comment line #auto_aliases=no openser advertises at startup: Aliases: tcp: sip.foo.bar:5060 udp: sip.foo.bar:5060 *: sip.foo.bar:5060:* and it DOES recognize that above Route header is local. if i uncomment auto_aliases=no and change alias="sip.foo.bar:5060" to alias="udp:sip.foo.bar:5060" alias="tcp:sip.foo.bar:5060" then loose_route DOES NOT recognize that the route header is local. my question is, how should i write the alias statements so that the route header is recognized local even when i have auto_aliases=no? -- juha

3 4

Re: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem
by Kostas Marneris 20 Mar '07

20 Mar '07

Hello, What I understand (from Frame12) Session Initiation Protocol Request-Line: BYE sip:000003@1.255.ua_priv__IP SIP/2.0 is that your GW does not comply with the fixed SIP UA Contact address Contact: <sip:000003@213.156.ua_pub_IP:1176> which SER seems to correctly fix it (from Frame6). GW must send the BYE request back to the fixed addr : 213.156.ua_pub_IP:1176. Try a different PSTN-GW or try to isolate the RTP-Proxy solution. I am really confused of your ser.cfg. You use force_rtp_proxy("l") on Loose Route Section / INVITE block, but you don't use force_rtp_proxy() on INVITE handler (Route[3]) and you don't also use rtp_proxy on BYE block at Loose Route Section. I tried this senario without Mediaproxy or RTPproxy with success. regards, Kostas --- K.Marneris(a)otenet.gr ----- Original Message ----- From: "Fabio Macchi" <f.macchi(a)keeptelecom.com> To: <greger(a)teigre.com> Cc: "'Kostas Marneris'" <K.Marneris(a)otenet.gr>; <serusers(a)lists.iptel.org> Sent: 20 March 2007 19:56 Subject: R: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hi Greger, I attached an ethereal SIP call trace of a test call ( summary and detailed, I simple maskerade final ip numbers ): below only the INVITE relayed from proxy to gateway: Session Initiation Protocol Request-Line: INVITE sip:9999001234@194.244.gatewayIP:5060 SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: <sip:194.244.Proxy__IP;ftag=12e1e2e19d527792;lr=on> Via: SIP/2.0/UDP 194.244.Proxy__IP;branch=z9hG4bKb24f.5133a2c4.0 Transport: UDP Sent-by Address: 194.244.Proxy__IP Branch: z9hG4bKb24f.5133a2c4.0 Via: SIP/2.0/UDP 1.255.ua_priv__IP;rport=1176;received=213.156.ua_pub_IP;branch=z9hG4bK35ca7a df63e3094f Transport: UDP Sent-by Address: 1.255.ua_priv__IP RPort: 1176 Received: 213.156.ua_pub_IP Branch: z9hG4bK35ca7adf63e3094f From: "000003" <sip:000003@194.244.Proxy__IP>;tag=12e1e2e19d527792 SIP Display info: "000003" SIP from address: sip:000003@194.244.Proxy__IP SIP tag: 12e1e2e19d527792 To: <sip:9999001234@194.244.Proxy__IP> SIP to address: sip:9999001234@194.244.Proxy__IP Contact: <sip:000003@213.156.ua_pub_IP:1176> Contact Binding: <sip:000003@213.156.ua_pub_IP:1176> URI: <sip:000003@213.156.ua_pub_IP:1176> SIP contact address: sip:000003@213.156.ua_pub_IP:1176 Supported: replaces Call-ID: 953e8996cfcc4ccc(a)1.255.ua_priv__IP CSeq: 60577 INVITE Sequence Number: 60577 Method: INVITE User-Agent: Grandstream HT386 1.0.3.64 FXS0 Max-Forwards: 16 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 327 Message body As you can see, contact informations are correctly fixed with the pubblic UA address, but when the callee hungs up, the BYE is relayed to the private IP address: am I missing something ? In this invite I see UA private address only in VIA: does BYE look to this parameter ? Later caller hangs up too, and the OK is relayed to the correct IP/port. Any help would be high appreciate, thanks in advance. Fabio _____ Da: Greger V. Teigre [mailto:greger@teigre.com] Inviato: martedμ 20 marzo 2007 9.59 A: Fabio Macchi Cc: 'Kostas Marneris'; serusers(a)lists.iptel.org Oggetto: Re: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Normally this happens because you haven't fixed the Contact if the original INVITE or OK. g-) Fabio Macchi wrote: First, thanks for answer. I've tryed your trik and in effect this solve the problem of the '200 ok' forwarded to the UA, but my problem still remain alive: when BYE is sent from Gateway, it reaches correctly SER, but it still forward it to the private UA address. I was wondering about the nat_uac_test in this case, as the source of the BYE message is the gateway ( not natted ) and not the UA. Have any idea about this ? Fabio -----Messaggio originale----- Da: Kostas Marneris [mailto:K.Marneris@otenet.gr] Inviato: giovedμ 15 marzo 2007 20.39 A: Fabio Macchi Cc: serusers(a)lists.iptel.org Oggetto: Re: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hello, I was working on about the same problem today either with 'Mediaproxy solution' or with 'SER's Nathelper only solution' . The NAT issue is a nightmare, not because of SER but because of different implementations on NAT boxes. Actually my problem was : if the NATed UA send a BYE to SER, SER forward it to PSTN-GW, then the '200 Ok' Response from PSTN-GW is forwarded by SER to UA to the wrong port (Contact or Via header port). I used the following block on Loose Route section, (because BYE is loose_routed if you use Record-Route), and it seems to work. # --------------------------------------- # Loose Route Section # --------------------------------------- if (loose_route()) { # mark routing logic in request if (method == "BYE") { if (nat_uac_test("22")) { xlog("L_NOTICE", "*** LR -> NATed BYE - Use force_rport()"); force_rport(); }; }; route(1); break; }; I faced up your second problem too. The solution was to move the NAT handling block before proxy_authorize block. I think that the different behaviour does not come with the 'standard RFC1918 addresses', but with the different NAT type. I realize that the provisional mesgs '100 Trying' and '407 Proxy Authentication Required' are relayed back to the real IP addr of NATed UA (this is correct), but to the WRONG port (that of Contact/Via header and not the signalling received port). It seems that these mesgs use the IP address part of 'Received' field of Location DB but not the port. It happens to work if NAT box use the SAME port (eg. 5060) on NAT translation (10.10.10.1:5060 --> Real_IP:5060) (eg. with a SAGEM1500 Router) But it does not work if NAT box doesn't use the same port (10.10.10.1:5060 --> Real_IP:38181) I think that this has to be verified by SER developers or SER experts. Kostas --- K.Marneris(a)otenet.gr ----- Original Message ----- From: "Fabio Macchi" <mailto:f.macchi@keeptelecom.com> <f.macchi(a)keeptelecom.com> To: <mailto:serusers@lists.iptel.org> <serusers(a)lists.iptel.org> Sent: 15 March 2007 19:34 Subject: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hi all, I'm running the following schema: UA ( possibly natted ) -> SER -> PSTN Gateway I have a problem with UA belonging to a particular network with private address not RFC1918 compliant ( class 1.x.x.x ), SER and PSTN Gateway have pubblic address. The problem is that, after a succesfull call, if the PSTN gateway send a BYE to SER, then SER forward BYE to the private address of UA instead of pubblic one. I don't understand which is the section that handle BYE messages and how can I solve this problem: anyone may help ? Second, another question: with this particular network I had problem with INVITE too, because SER was sending "proxy authorization request" to the wrong TCP port. To solve this, I've moved the nat handling ( with force_rport ) before the proxy_authorize block and it's working, but why this is not necessary on standard RFC1918 compliant natted address ? Many thanks for any explanation Fabio _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------------------- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.15/728 - Release Date: 20/03/2007 08:07

1 0

Re: R: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem
by Greger V. Teigre 20 Mar '07

20 Mar '07

Thanks for the detailed trace, I wouldn't have spent on this without it ;-) Its' the ACK for the OK that is not done fix_nated_contact for. In fact, checking the getting started rtpproxy file, it seems that this error is also found there. All messages coming from behind NAT should have the contact fixed. Strictly speaking, RFC3261 states that the dialog state should be created from the information found in the INVITE, not the ACK on an OK. Your BYE generating entity does this error. I have seen this problem before, but has not really thought about it from the getting started config file perspective. It is thus safer to change Contact on ALL messages behind NAT, regardless of they are loose routed or not. g-) Fabio Macchi wrote: > > Hi Greger, > > > > I attached an ethereal SIP call trace of a test call ( summary and > detailed, I simple maskerade final ip numbers ): below only the INVITE > relayed from proxy to gateway: > > > > Session Initiation Protocol > > Request-Line: INVITE sip:9999001234@194.244.gatewayIP:5060 SIP/2.0 > > Method: INVITE > > [Resent Packet: False] > > Message Header > > Record-Route: <sip:194.244.Proxy__IP;ftag=12e1e2e19d527792;lr=on> > > Via: SIP/2.0/UDP 194.244.Proxy__IP;branch=z9hG4bKb24f.5133a2c4.0 > > Transport: UDP > > Sent-by Address: 194.244.Proxy__IP > > Branch: z9hG4bKb24f.5133a2c4.0 > > Via: SIP/2.0/UDP > 1.255.ua_priv__IP;rport=1176;received=213.156.ua_pub_IP;branch=z9hG4bK35ca7adf63e3094f > > Transport: UDP > > Sent-by Address: 1.255.ua_priv__IP > > RPort: 1176 > > Received: 213.156.ua_pub_IP > > Branch: z9hG4bK35ca7adf63e3094f > > From: "000003" <sip:000003@194.244.Proxy__IP>;tag=12e1e2e19d527792 > > SIP Display info: "000003" > > SIP from address: sip:000003@194.244.Proxy__IP > > SIP tag: 12e1e2e19d527792 > > To: <sip:9999001234@194.244.Proxy__IP> > > SIP to address: sip:9999001234@194.244.Proxy__IP > > Contact: <sip:000003@213.156.ua_pub_IP:1176> > > Contact Binding: <sip:000003@213.156.ua_pub_IP:1176> > > URI: <sip:000003@213.156.ua_pub_IP:1176> > > SIP contact address: sip:000003@213.156.ua_pub_IP:1176 > > Supported: replaces > > Call-ID: 953e8996cfcc4ccc(a)1.255.ua_priv__IP > > CSeq: 60577 INVITE > > Sequence Number: 60577 > > Method: INVITE > > User-Agent: Grandstream HT386 1.0.3.64 FXS0 > > Max-Forwards: 16 > > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE > > Content-Type: application/sdp > > Content-Length: 327 > > Message body > > > > As you can see, contact informations are correctly fixed with the > pubblic UA address, but when the callee hungs up, the BYE is relayed > to the private IP address: am I missing something ? > > In this invite I see UA private address only in VIA: does BYE look to > this parameter ? > > Later caller hangs up too, and the OK is relayed to the correct IP/port. > > > > Any help would be high appreciate, thanks in advance. > > > > Fabio > > > >

1 0

[Users] NAT Problem with OpenSER behind load balancer
by Tim Madorma 20 Mar '07

20 Mar '07

Hi, I have put the OpenSER version 1.2 behind a BIG IP load balancer which provides a static private IP address (10.1.1.20) via NAT to the OpenSER. In our application, the clients will send to one of the three public addresses that are defined as aliases: alias="65.185.232.62:5061" alias="65.185.232.62:5062" alias="65.185.232.62:5063" An incoming OPTIONS message is received with the host in the REQ URI set to 65.185.232.62:5063 is received and when it hits this section of openser.cfg: if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; it executes within the if conditional and tries to route the OPTIONS message. I am wondering why uri does not match myself. The debug output from the log is as follows (it seems that the alias assignments are not working): Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 0.1.1.20] Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] grep_sock_info - checking if port 5061 matches port 5063 Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 0.1.1.20] Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] grep_sock_info - checking if port 5062 matches port 5063 Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 0.1.1.20] Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] grep_sock_info - checking if port 5063 matches port 5063 Mar 19 16:41:11 homer openser[2267]: [ID 140248 local0.debug] DEBUG:check_self: host != me Any suggestions? I have not tried this in any other version. Is this a possible bug in 1.2? thanks, Tim

3 2

RV: RV: [Users] WeSIP session question
by Antonio Abajo 20 Mar '07

20 Mar '07

Buenas tardes Tele In the capture, I see the following call flow. GW WESIP MS ------ INV(S1)-----> -------INV (S2) ------> <--------183 ----------- ------- 183 -----------> <----------ACK --------- <---------503 ---------- ... 14:07:57 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Request Received <<<<<<<<< INVITE sip:390104491079@82.215.163.67 SIP/2.0 Via: SIP/2.0/UDP 82.215.163.5:5060;branch=z9hG4bK6a4fa6b1 Max-Forwards: 69 From: <sip:3405300695@82.215.163.5>;tag=6a4fa6b1 To: <sip:390104491079@82.215.163.67> Call-ID: 1783604896-30787@SVIGateway CSeq: 1 INVITE Contact: <sip:82.215.163.5:5060> Content-Type: application/sdp Content-Length: 213 14:07:58 19Mar2007 DEBUG SipRequest [SipProcessor[4]]- >>>>>>>>> Sending Request >>>>>>>>> INVITE sip:199@82.215.133.50 SIP/2.0 Max-Forwards: 69 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A To: <sip:390104491079@82.215.163.67> CSeq: 1 INVITE Content-Type: application/sdp Call-ID: 11111783604896-30787@SVIGateway Contact: <sip:82.215.163.67:5060;transport=udp> Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439 Content-Length: 21314:07:58 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Response Received <<<<<<<<< SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58 Date: Mon, 25 Mar 2002 04:04:50 GMT Call-ID: 11111783604896-30787@SVIGateway Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,UPDATE,REGISTER,SUBSCRIBE,NOTIFY,PRACK,RE FER Allow-Events: telephone-event Contact: <sip:199@82.215.133.50:5060> Content-Type: application/sdp Content-Disposition: oSystemsSIP-GW-UserAg Content-Length: 235 14:07:58 19Mar2007 DEBUG SipResponse [SipProcessor[4]]- >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 82.215.163.5:5060;branch=z9hG4bK6a4fa6b1 Max-Forwards: 69 From: <sip:3405300695@82.215.163.5>;tag=6a4fa6b1 To: <sip:390104491079@82.215.163.67> CSeq: 1 INVITE Call-ID: 11111783604896-30787@SVIGateway Content-Type: application/sdp Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 9756 1908 IN IP4 82.215.133.50 s=SIP Call c=IN IP4 82.215.133.50 t=0 0 m=audio 19450 RTP/AVP 0 99 c=IN IP4 82.215.133.50 a=rtpmap:0 PCMU/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 14:08:07 19Mar2007 DEBUG SipProcessor [SipProcessor[3]]- <<<<<<<<< Request Received <<<<<<<<< ACK sip:199@82.215.133.50 SIP/2.0 Via: SIP/2.0/UDP 82.215.163.67;branch=z9hG4bK07de.99a696f6.0 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A Call-ID: 11111783604896-30787@SVIGateway To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58 CSeq: 1 ACK User-Agent: OpenSer (1.2.0-pre6-notls (i386/linux)) Content-Length: 0 14:08:07 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Response Received <<<<<<<<< SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58 Date: Mon, 25 Mar 2002 04:04:50 GMT Call-ID: 11111783604896-30787@SVIGateway Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Reason: Q.850;cause=41 Content-Length: 0 The signalling is not correct but, I have looked your code and believe that it is right, revise it and you check if you have the last version of WeSIP and you return to try, because the last release commited in HYPERLINK "http://www.wesip.com/mediawiki/index.php/Downloads:WeSIP_Beta_0.1"http://ww w.wesip.com/mediawiki/index.php/Downloads:WeSIP_Beta_0.1 was fixed a bug with the ApplicationSessions. In my test application I reply your behaviour(with 5xx, with AppSession attributes) and all have working perfectly, I dont understand nothing :( Sorry Good Luck. Antonio -----Mensaje original----- De: tele [mailto:tele@plexia.com] Enviado el: martes, 20 de marzo de 2007 16:46 Para: users(a)openser.org CC: antonio.abajo(a)voztele.com Asunto: Re: RV: [Users] WeSIP session question Hola Antonio, Your example works great and the code is more clear and correct than mine :) i was not clear, my problem was only with the 503 Service Unavailable response, that means a 5xx Server Failure, is it possibile that when the container or openser with seas recive a 5xx response is not able to do for example a creation of a new invite to UA3 recover from the same session?. Or probably i'm occur in a 473 response for an incorrect use of SipApplicationSession. anyway with a 480 Temporarily Unavailable response from the UA2 i solve my problem. thank you very much :tele On Mon, 2007-03-19 at 19:27 +0100, Antonio Abajo wrote: > Hola tele ;) > > I have tested an example similar to yours, and it works ok. It's de > following: > > UA1 ---[1]--> WeSIP --[2]--486 Busy Here---> UA2 > | > [3] > ---OK----_> UA3 > > [1] UA1 generates an INVITE to UA2 ant it 's forwarded to UA2 with a B2BUA > behaviour. In do INVITE. I have implemented the following: > > protected void doInvite(SipServletRequest invite){ > SipServletRequest otherInvite = sf.createRequest(invite, false); > SipURI sipUri = sf.createSipURI("UA2","proxy"); > otherInvite.setRequestURI(sipUri); > otherInvite.getSession().setAttribute("REQUEST", otherInvite); > > otherInvite.getSession().setAttribute("PEER_SESSION",invite.getSession()); > > invite.getSession().setAttribute("PEER_SESSION",otherInvite.getSession()); > invite.getSession().setAttribute("REQUEST", invite); > otherInvite.send(); > } > > [2] UA2 declines the call and a 486 response is received by WeSIP and > process it in doErrorResponse: > > protected void doErrorResponse(SipServletResponse errorResponse) { > switch(errorResponse.getStatus()){ > case 486: > SipServletRequest request = (SipServletRequest) > errorResponse.getSession().getAttribute("REQUEST"); > SipSession otherSession = (SipSession) > errorResponse.getSession().getAttribute("PEER_SESSION"); > SipURI sipUri = sf.createSipURI("UA3", "proxy"); > request.setRequestURI(sipUri); > SipServletRequest newRequest = > sf.createRequest(request,false); > newRequest.getSession().setAttribute("REQUEST", newRequest); > newRequest.getSession().setAttribute("PEER_SESSION", > otherSession); > otherSession.setAttribute("PEER_SESSION", > newRequest.getSession()); > newRequest.setHeader("X-SSVTPBX", "yes"); > newRequest.send(); > break; > } > } > > [3] UA3 Recevies the second call and it takes down. > > > I do not understand the cause of error. I annex you the code of simple sip > application example that replies to your problem. You can start it up and > checking if it works. Else if work for you, I would attempt watching the > particular case when a 5XX responses are received. > > Sorry... > > Best regards. > Antonio. > > -----Mensaje original----- > De: tele [mailto:tele@plexia.com] > Enviado el: lunes, 19 de marzo de 2007 14:33 > Para: Antonio Abajo Álvarez > Asunto: RE: [Users] WeSIP session question > > Hola Antonio, > > The scenario is more complex, i'll try to explain it: > > PSTN > | > | > UA---->GW ------> WeSIP(B2BUA) > | > | > mediaserver > > > GW: 82.215.163.5 > WeSIP: 82.215.163.67 > MS: 82.215.133.50 > > in the mediaserver there is an vxml script that i play in early media > and in case of particular event return to wesip a 503 temporaly > unavailable or e 410 Gone. So my B2BUA application have control of this > and can do stuff with the 503 and 410. > > in particular, in case of 503 temporaly unavailable i get the upstream > session and create a new invite to the media server for play another > announcement associated. in case of 410 gone i generate a new invite to > the GW with the original URI request for the correct termination. > > Yes when the session is removed i'm able to send another call. > > attached here the logs and the servlet. > > don't care about the hardcoded IP and the repeated code :-) > i'm doing testing.. > > regards, > > :tele > > > On Mon, 2007-03-19 at 13:29 +0100, Antonio Abajo Álvarez wrote: > > Hi Tele, > > > > I don't understand very well the problem., for what I understand you have > > the following: > > > > > > UA1 --------------------- WeSIP --------------------- UA2 > > > > ---INV/4XX-6XX/ACK (SS1) --> ----INV/4XX-6XX/ACK(SS2)--> > > > > You try to send another call and receive the 473 response of WeSIP... > > > > ------INV/473/ACK (SS1) --> > > > > And when the session has been removed you can send another call. > > > > If it is the case, I understand that the 473 response is send from > > application or from openser script configuration, because the internal > > behaviour of SIP doesn't send this response automatically. > > > > Can you verify if it is the case? > > > > Thank you very much... > > > > Antonio. > > > > -----Mensaje original----- > > De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre > > de tele > > Enviado el: lunes, 19 de marzo de 2007 12:35 > > Para: users(a)openser.org > > Asunto: [Users] WeSIP session question > > > > Hi, > > > > I've a problem with WeSIP in B2BUA mode, in case of failed call 4xx-6xx > > correctly terminated, when i try to send another call to WeSIP i recive > > a "473 Filtered destination" then for send another call i've to wait > > WeSIP complete some management with session: > > > > 14:10:16 19Mar2007 DEBUG SipConnector [SipProcessor[3]]- recycle: > > Recycling processor SipProcessor[3] > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e6006f] in state [3] with lifetime of :74490 > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e68616] in state [3] with lifetime of :40281 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e6006f] in state [3] with lifetime of :134500 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e68616] in state [3] with lifetime of :100291 > > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =0 > > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- Remove AppSession > > [B4D6C9C4288784A68E032D20AB78BD8E] > > > > When i see Remove AppSession i'm able to send another call... > > I've read the sip servlet spec about that and i'm trying to invalidate() > > or setExpires() to SiApplicationSession in case of failed call. but it's > > not clear how to do yet. > > > > i can provide the full debug if needed. > > > > thank you! > > > > regards > > > > :tele > > > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49

1 0

[Users] DB migration
by Leonid Fainshtein 20 Mar '07

20 Mar '07

Hi, I didn't understand the proposed process of MySQL DB migration from ver.1.1 to 1.2 For example, the result of "openser_mysqldb.sh migrate openser openser12" will be a new database openser12. But how can I copy/rename openser12 to openser? Does it mean that from now, all of my auxiliary software must work with openser12? Thank you, Leonid Fainshtein ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

2 1

[Users] Help with dbtext location table - Almost there!
by Daryl Sanders 20 Mar '07

20 Mar '07

I have been struggling to get usrloc to work with... modparam("usrloc", "db_mode", 1) OpenSER 1.2 starts fine but gives the following error when a phone tries to register: 2(26634) db_insert_ucontact(): Error while inserting contact 2(26634) ERROR:usrloc:insert_ucontact: failed to insert in database Regards, Daryl

1 0

RV: [Users] WeSIP session question
by Antonio Abajo 20 Mar '07

20 Mar '07

Hola tele ;) I have tested an example similar to yours, and it works ok. It's de following: UA1 ---[1]--> WeSIP --[2]--486 Busy Here---> UA2 | [3] ---OK----_> UA3 [1] UA1 generates an INVITE to UA2 ant it 's forwarded to UA2 with a B2BUA behaviour. In do INVITE. I have implemented the following: protected void doInvite(SipServletRequest invite){ SipServletRequest otherInvite = sf.createRequest(invite, false); SipURI sipUri = sf.createSipURI("UA2","proxy"); otherInvite.setRequestURI(sipUri); otherInvite.getSession().setAttribute("REQUEST", otherInvite); otherInvite.getSession().setAttribute("PEER_SESSION",invite.getSession()); invite.getSession().setAttribute("PEER_SESSION",otherInvite.getSession()); invite.getSession().setAttribute("REQUEST", invite); otherInvite.send(); } [2] UA2 declines the call and a 486 response is received by WeSIP and process it in doErrorResponse: protected void doErrorResponse(SipServletResponse errorResponse) { switch(errorResponse.getStatus()){ case 486: SipServletRequest request = (SipServletRequest) errorResponse.getSession().getAttribute("REQUEST"); SipSession otherSession = (SipSession) errorResponse.getSession().getAttribute("PEER_SESSION"); SipURI sipUri = sf.createSipURI("UA3", "proxy"); request.setRequestURI(sipUri); SipServletRequest newRequest = sf.createRequest(request,false); newRequest.getSession().setAttribute("REQUEST", newRequest); newRequest.getSession().setAttribute("PEER_SESSION", otherSession); otherSession.setAttribute("PEER_SESSION", newRequest.getSession()); newRequest.setHeader("X-SSVTPBX", "yes"); newRequest.send(); break; } } [3] UA3 Recevies the second call and it takes down. I do not understand the cause of error. I annex you the code of simple sip application example that replies to your problem. You can start it up and checking if it works. Else if work for you, I would attempt watching the particular case when a 5XX responses are received. Sorry... Best regards. Antonio. -----Mensaje original----- De: tele [mailto:tele@plexia.com] Enviado el: lunes, 19 de marzo de 2007 14:33 Para: Antonio Abajo Álvarez Asunto: RE: [Users] WeSIP session question Hola Antonio, The scenario is more complex, i'll try to explain it: PSTN | | UA---->GW ------> WeSIP(B2BUA) | | mediaserver GW: 82.215.163.5 WeSIP: 82.215.163.67 MS: 82.215.133.50 in the mediaserver there is an vxml script that i play in early media and in case of particular event return to wesip a 503 temporaly unavailable or e 410 Gone. So my B2BUA application have control of this and can do stuff with the 503 and 410. in particular, in case of 503 temporaly unavailable i get the upstream session and create a new invite to the media server for play another announcement associated. in case of 410 gone i generate a new invite to the GW with the original URI request for the correct termination. Yes when the session is removed i'm able to send another call. attached here the logs and the servlet. don't care about the hardcoded IP and the repeated code :-) i'm doing testing.. regards, :tele On Mon, 2007-03-19 at 13:29 +0100, Antonio Abajo Álvarez wrote: > Hi Tele, > > I don't understand very well the problem., for what I understand you have > the following: > > > UA1 --------------------- WeSIP --------------------- UA2 > > ---INV/4XX-6XX/ACK (SS1) --> ----INV/4XX-6XX/ACK(SS2)--> > > You try to send another call and receive the 473 response of WeSIP... > > ------INV/473/ACK (SS1) --> > > And when the session has been removed you can send another call. > > If it is the case, I understand that the 473 response is send from > application or from openser script configuration, because the internal > behaviour of SIP doesn't send this response automatically. > > Can you verify if it is the case? > > Thank you very much... > > Antonio. > > -----Mensaje original----- > De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre > de tele > Enviado el: lunes, 19 de marzo de 2007 12:35 > Para: users(a)openser.org > Asunto: [Users] WeSIP session question > > Hi, > > I've a problem with WeSIP in B2BUA mode, in case of failed call 4xx-6xx > correctly terminated, when i try to send another call to WeSIP i recive > a "473 Filtered destination" then for send another call i've to wait > WeSIP complete some management with session: > > 14:10:16 19Mar2007 DEBUG SipConnector [SipProcessor[3]]- recycle: > Recycling processor SipProcessor[3] > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e6006f] in state [3] with lifetime of :74490 > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e68616] in state [3] with lifetime of :40281 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e6006f] in state [3] with lifetime of :134500 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e68616] in state [3] with lifetime of :100291 > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =0 > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- Remove AppSession > [B4D6C9C4288784A68E032D20AB78BD8E] > > When i see Remove AppSession i'm able to send another call... > I've read the sip servlet spec about that and i'm trying to invalidate() > or setExpires() to SiApplicationSession in case of failed call. but it's > not clear how to do yet. > > i can provide the full debug if needed. > > thank you! > > regards > > :tele > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.13/726 - Release Date: 18/03/2007 15:34

2 1

[Users] mi_fifo question
by Andreas Granig 20 Mar '07

20 Mar '07

Hi, I have a OpenSER 1.2 with mi_fifo running, but adding an alias fails: /usr/local/openser-1.2# ./sbin/openserctl alias add 1000 sip:foo1000@mydomain.org database engine 'MYSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 entering fifo_cmd ul_add aliases 1000(a)mydomain.org sip:foo1000@mydomain.org 0 1.00 0 0 4294967295 400 Too few or too many arguments FIFO command was: :ul_add:openser_receiver_17638 aliases 1000(a)mydomain.org sip:foo1000@mydomain.org 0 1.00 0 0 4294967295 In etc/openser/openserctlrc I have ALIASES_TYPE="UL", and the mi_fifo module config looks like this: 88 loadmodule "mi_fifo.so" 89 modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") 90 modparam("mi_fifo", "fifo_mode", 0660) 91 modparam("mi_fifo", "fifo_group", "openser") 92 modparam("mi_fifo", "fifo_user", "openser") 93 modparam("mi_fifo", "reply_dir", "/tmp/") 94 modparam("mi_fifo", "reply_indent", "\t") Am I missing something somewhere? Cheers, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

3 3

[Users] loose_route problem with 1.2
by jh＠tutpro.com 20 Mar '07

20 Mar '07

i have in openser.cfg file listen=x.x.x.x port=5060 alias="sip.test.fi:5060" tcp_accept_aliases=yes out-of-dialog request comes in with pre-loaded route header: Route: <sip:sip.test.fi;lr;transport=TCP> loose_route() returns true although according to rr/readme it should return false: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE. this is with openser 1.2. i don't remember having this problems with openser 1.1.1. has something changed in config or what it is that i'm missing? -- juha

2 1

[Serusers] Find an issue with FIFO after crash
by inge 20 Mar '07

20 Mar '07

Hi all, After a crash, the server runs normally SER process. In /tmp/ I've my ser_fifo with same rights than an other server. But my SERWeb says : Warning: filetype(): Lstat failed for (null) (errno=13 - Permission denied) in /var/www/client/functions.php on line 332 FIFO not running or bad path to it, file: /var/www/client/functions.php:335 I've make a test by extracting the if bloc in functions.php near line 332. I put this in an other file. When I execute "php test.php", filetype is Okay. But when I access by http://.../test.php I've got the same error than serweb. Thanks for your support. Adrien

2 4

[Serusers] Unable to compile the xmlrpc module
by Fabian Dedu 20 Mar '07

20 Mar '07

Hi all, I'm trying to compile the ser-2.0.0+cvs20070315_src.tar.gz version and I'm obtaining the following error when trying to compile with xmlrpc support: Input: make include_modules="mysql xmlrpc" all ------------------------------------------------------------------------------------------------------------ make[1]: Entering directory `/downloads/ser2.0/ser/modules/mysql' make[1]: `mysql.so' is up to date. make[1]: Leaving directory `/downloads/ser2.0/ser/modules/mysql' make[1]: Entering directory `/downloads/ser2.0/ser/modules/xmlrpc' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -mcpu=athlon -Wall -DNAME='"ser"' -DVERSION='" 2.0.0+cvs20070309"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 3.2.3"' -D__CPU_i386 -D__OS_linux -DSER_VER=2000000 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -I/usr/include/libxml2 -I/usr/local/include/libxml2 -I/usr/local/include -c xmlrpc.c -o xmlrpc.o xmlrpc.c: In function `open_doc': xmlrpc.c:1159: warning: implicit declaration of function `xmlReadMemory' xmlrpc.c:1160: `XML_PARSE_NOBLANKS' undeclared (first use in this function) xmlrpc.c:1160: (Each undeclared identifier is reported only once xmlrpc.c:1160: for each function it appears in.) xmlrpc.c:1161: `XML_PARSE_NONET' undeclared (first use in this function) xmlrpc.c:1162: `XML_PARSE_NOCDATA' undeclared (first use in this function) xmlrpc.c:1162: warning: assignment makes pointer from integer without a cast xmlrpc.c: In function `select_method': xmlrpc.c:1505: `XML_PARSE_NOBLANKS' undeclared (first use in this function) xmlrpc.c :1505: `XML_PARSE_NONET' undeclared (first use in this function) xmlrpc.c:1505: `XML_PARSE_NOCDATA' undeclared (first use in this function) xmlrpc.c:1505: warning: assignment makes pointer from integer without a cast make[1]: *** [xmlrpc.o] Error 1 make[1]: Leaving directory `/downloads/ser2.0/ser/modules/xmlrpc' make[1]: Entering directory `/downloads/ser2.0/ser/modules/acc_db' make[1]: `acc_db.so' is up to date. make[1]: Leaving directory `/downloads/ser2.0/ser/modules/acc_db' etc.... all the other modules are compiling okay ------------------------------------------------------------------------------------------------------------ I'm using Red Hat Enterprise AS 3.0 with all updates included. Did anyone experiencing the above compile issue regarding this new xmlrpc module ? Thank you, Fabian.

2 4

[Serusers] SER -> PSTN Gateway+NAT: BYE handling problem
by Fabio Macchi 20 Mar '07

20 Mar '07

Hi all, I'm running the following schema: UA ( possibly natted ) -> SER -> PSTN Gateway I have a problem with UA belonging to a particular network with private address not RFC1918 compliant ( class 1.x.x.x ), SER and PSTN Gateway have pubblic address. The problem is that, after a succesfull call, if the PSTN gateway send a BYE to SER, then SER forward BYE to the private address of UA instead of pubblic one. I don't understand which is the section that handle BYE messages and how can I solve this problem: anyone may help ? Second, another question: with this particular network I had problem with INVITE too, because SER was sending "proxy authorization request" to the wrong TCP port. To solve this, I've moved the nat handling ( with force_rport ) before the proxy_authorize block and it's working, but why this is not necessary on standard RFC1918 compliant natted address ? Many thanks for any explanation Fabio

3 3

[Serusers] Heartbeat with ser and asterisk
by aespinoza＠vivophone.com 20 Mar '07

20 Mar '07

Hello people I have a heartbeat cluster that manages a ser 0.9.3 running on one machine and asterisk1.2.3 running on another in an Active/Active two IP address Configuration with failover support. I have SUSE 10.1 installed on both machines. My ha.cf files look like this ############################################### logfile /var/log/ha-log logfacility local0 keepalive 2 deadtime 10 warntime 10 initdead 20 udpport 694 baud 19200 bcast eth1 ping xxx.xxx.x.x auto_failback on node linux-xczz node prueba2 respawn hacluster /usr/local/lib/heartbeat/ipfail ################################################# My haresources files look like this ######################################## prueba2 xxx.xxx.x.125/24 safe_asterisk linux-xczz xxx.xxx.x.124/24 serctl ######################################## so "linux-xczz" is the master when running ser and "prueba2" is the master when running asterisk the authkeys files are the same on both machines too with the right permissions (mod 600). The /etc/hosts files look like this ########################## 10.10.10.1 linux-xczz 10.10.10.2 prueba2 ########################## The First time I try to run heartbeat on one machine (prueba2) with /etc/init.d/heartbeat start, both my services run good. But when I try to run heartbeat on the other machine (linux-xczz) so that it takes over the ser service, the system goes crazy and once linux-xczz takes over the ser service, prueba2 gives up the other resource (asterisk) which it should not do, and it appears on linux-xczz, only to disappear seconds later along with ser, leaving my cluster-ha a complete wreck with no service running on either machine. The error log I get from prueba2 is this: heartbeat[12609]: 2007/03/15_11:41:18 info: Link linux-xczz:eth1 up. heartbeat[12609]: 2007/03/15_11:41:18 info: Status update for node linux-xczz: status init heartbeat[12609]: 2007/03/15_11:41:18 info: Status update for node linux-xczz: status up harc[13730]: 2007/03/15_11:41:18 info: Running /etc/ha.d/rc.d/status status harc[13741]: 2007/03/15_11:41:18 info: Running /etc/ha.d/rc.d/status status heartbeat[12609]: 2007/03/15_11:41:19 info: Status update for node linux-xczz: status active harc[13754]: 2007/03/15_11:41:19 info: Running /etc/ha.d/rc.d/status status heartbeat[12609]: 2007/03/15_11:41:19 info: remote resource transition completed. heartbeat[12609]: 2007/03/15_11:41:19 info: prueba2 wants to go standby [foreign] heartbeat[12609]: 2007/03/15_11:41:20 info: standby: linux-xczz can take our foreign resources heartbeat[13767]: 2007/03/15_11:41:20 info: give up foreign HA resources (standby). ResourceManager[13777]: 2007/03/15_11:41:20 info: Releasing resource group: linux-xczz xxx.xxx.x.124/24 serctl ResourceManager[13777]: 2007/03/15_11:41:20 info: Running /etc/init.d/serctl stop ResourceManager[13777]: 2007/03/15_11:41:20 info: Running /etc/ha.d/resource.d/IPaddr xxx.xxx.x.124/24 stop IPaddr[13915]: 2007/03/15_11:41:20 INFO: /sbin/route -n del -host xxx.xxx.x.124 IPaddr[13915]: 2007/03/15_11:41:20 INFO: /sbin/ifconfig eth0:0 xxx.xxx.x.124 down IPaddr[13915]: 2007/03/15_11:41:20 INFO: IP Address xxx.xxx.x.124 released IPaddr[13836]: 2007/03/15_11:41:20 INFO: IPaddr Success heartbeat[13767]: 2007/03/15_11:41:20 info: foreign HA resource release completed (standby). heartbeat[12609]: 2007/03/15_11:41:20 info: Local standby process completed [foreign]. heartbeat[12609]: 2007/03/15_11:41:23 WARN: 1 lost packet(s) for [linux-xczz] [13:15] heartbeat[12609]: 2007/03/15_11:41:23 info: remote resource transition completed. heartbeat[12609]: 2007/03/15_11:41:23 info: No pkts missing from linux-xczz! heartbeat[12609]: 2007/03/15_11:41:23 info: Other node completed standby takeover of foreign resources. heartbeat[12609]: 2007/03/15_11:41:35 info: linux-xczz wants to go standby [foreign] heartbeat[12609]: 2007/03/15_11:41:36 info: standby: acquire [foreign] resources from linux-xczz heartbeat[14011]: 2007/03/15_11:41:36 info: acquire local HA resources (standby). ResourceManager[14021]: 2007/03/15_11:41:36 info: Acquiring resource group: prueba2 xxx.xxx.x.125/24 asterisk-rosa IPaddr[14048]: 2007/03/15_11:41:36 INFO: IPaddr Running OK ResourceManager[14021]: 2007/03/15_11:41:36 info: Running /etc/init.d/safe_asterisk start ResourceManager[14021]: 2007/03/15_11:41:36 ERROR: Return code 1 from /etc/init.d/safe_asterisk ResourceManager[14021]: 2007/03/15_11:41:36 CRIT: Giving up resources due to failure of safe_asterisk ResourceManager[14021]: 2007/03/15_11:41:36 info: Releasing resource group: prueba2 xxx.xxx.x.125/24 asterisk-rosa ResourceManager[14021]: 2007/03/15_11:41:xxz.xxz.x.xxz36 info: Running /etc/init.d/safe_asterisk stop ResourceManager[14021]: 2007/03/15_11:41:37 info: Running /etc/ha.d/resource.d/IPaddr xxx.xxx.x.125/24 stop IPaddr[14310]: 2007/03/15_11:41:37 INFO: /sbin/route -n del -host xxx.xxx.x.125 IPaddr[14310]: 2007/03/15_11:41:37 INFO: /sbin/ifconfig eth0:2 xxx.xxx.x.125 down IPaddr[14310]: 2007/03/15_11:41:37 INFO: IP Address xxx.xxx.x.125 released IPaddr[14231]: 2007/03/15_11:41:37 INFO: IPaddr Success heartbeat[14011]: 2007/03/15_11:41:37 info: local HA resource acquisition completed (standby). heartbeat[12609]: 2007/03/15_11:41:37 info: Standby resource acquisition done [foreign]. heartbeat[12609]: 2007/03/15_11:41:37 info: remote resource transition completed. heartbeat[12609]: 2007/03/15_11:41:38 WARN: G_CH_dispatch_int: Dispatch function for read child took too long to execute: 520 ms (> 50 ms) (GSource: 0x80fbe00) hb_standby[14375]: 2007/03/15_11:42:07 Going standby [foreign]. heartbeat[12609]: 2007/03/15_11:42:07 info: prueba2 wants to go standby [foreign] heartbeat[12609]: 2007/03/15_11:42:08 info: standby: linux-xczz can take our foreign resources heartbeat[14385]: 2007/03/15_11:42:08 info: give up foreign HA resources (standby). ResourceManager[14395]: 2007/03/15_11:42:08 info: Releasing resource group: linux-xczz xxx.xxx.x.124/24 serctl ResourceManager[14395]: 2007/03/15_11:42:08 info: Running /etc/init.d/serctl stop ResourceManager[14395]: 2007/03/15_11:42:08 ERROR: Return code 1 from /etc/init.d/serctl The error log I get from linux-xczz when I run heartbeat is this: heartbeat[1063]: 2007/03/15_14:26:12 WARN: Core dumps could be lost if multiple dumps occur heartbeat[1063]: 2007/03/15_14:26:12 WARN: Consider setting /proc/sys/kernel/core_uses_pid (or equivalent) to 1 for maximum supportability heartbeat[1063]: 2007/03/15_14:26:12 WARN: Logging daemon is disabled --enabling logging daemon is recommended heartbeat[1063]: 2007/03/15_14:26:12 info: ************************** heartbeat[1063]: 2007/03/15_14:26:12 info: Configuration validated. Starting heartbeat 2.0.7 heartbeat[1064]: 2007/03/15_14:26:12 info: heartbeat: version 2.0.7 heartbeat[1064]: 2007/03/15_14:26:12 info: Heartbeat generation: 130 heartbeat[1064]: 2007/03/15_14:26:12 info: G_main_add_TriggerHandler: Added signal manual handler heartbeat[1064]: 2007/03/15_14:26:12 info: G_main_add_TriggerHandler: Added signal manual handler heartbeat[1064]: 2007/03/15_14:26:12 info: Removing /usr/local/var/run/heartbeat/rsctmp failed, recreating. heartbeat[1064]: 2007/03/15_14:26:12 info: glib: UDP Broadcast heartbeat started on port 694 (694) interface eth1 heartbeat[1064]: 2007/03/15_14:26:12 info: glib: UDP Broadcast heartbeat closed on port 694 interface eth1 - Status: 1 heartbeat[1064]: 2007/03/15_14:26:12 info: glib: ping heartbeat started. heartbeat[1064]: 2007/03/15_14:26:12 info: G_main_add_SignalHandler: Added signal handler for signal 17 heartbeat[1064]: 2007/03/15_14:26:12 info: Local status now set to: 'up' heartbeat[1064]: 2007/03/15_14:26:13 info: Link linux-xczz:eth1 up. heartbeat[1064]: 2007/03/15_14:26:13 info: Link prueba2:eth1 up. heartbeat[1064]: 2007/03/15_14:26:13 info: Status update for node prueba2: status active heartbeat[1064]: 2007/03/15_14:26:13 info: Link xxx.xxx.x.x:xxx.xxx.x.x up. heartbeat[1064]: 2007/03/15_14:26:13 info: Status update for node xxx.xxx.x.x: status ping harc[1073]: 2007/03/15_14:26:13 info: Running /usr/local/etc/ha.d/rc.d/status status heartbeat[1064]: 2007/03/15_14:26:14 info: Comm_now_up(): updating status to active heartbeat[1064]: 2007/03/15_14:26:14 info: Local status now set to: 'active' heartbeat[1064]: 2007/03/15_14:26:14 info: Starting child client "/usr/local/lib/heartbeat/ipfail" (1001,100) heartbeat[1084]: 2007/03/15_14:26:14 info: Starting "/usr/local/lib/heartbeat/ipfail" as uid 1001 gid 100 (pid 1084) heartbeat[1064]: 2007/03/15_14:26:14 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:26:14 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:26:14 info: Local Resource acquisition completed. (none) heartbeat[1064]: 2007/03/15_14:26:15 info: prueba2 wants to go standby [foreign] heartbeat[1064]: 2007/03/15_14:26:15 info: standby: acquire [foreign] resources from prueba2 heartbeat[1088]: 2007/03/15_14:26:15 info: acquire local HA resources (standby). ResourceManager[1098]: 2007/03/15_14:26:15 info: Acquiring resource group: linux-xczz xxx.xxx.x.124/24 serctl IPaddr[1122]: 2007/03/15_14:26:16 INFO: IPaddr Resource is stopped ResourceManager[1098]: 2007/03/15_14:26:16 info: Running /usr/local/etc/ha.d/resource.d/IPaddr 192.168.1.124/24 start IPaddr[1321]: 2007/03/15_14:26:16 INFO: eval /sbin/ifconfig eth0:0 xxx.xxx.x.124 netmask 255.255.255.0 broadcast xxx.xxx.x.255 IPaddr[1321]: 2007/03/15_14:26:16 INFO: Sending Gratuitous Arp for xxx.xxx.x.124 on eth0:0 [eth0] IPaddr[1321]: 2007/03/15_14:26:16 INFO: /usr/local/lib/heartbeat/send_arp -i 500 -r 10 -p /usr/local/var/run/heartbeat/rsctmp/send_arp/send_arp-xxx.xxx.x.124 eth0 xxx.xxx.x.124 auto xxx.xxx.x.124 ffffffffffff IPaddr[1241]: 2007/03/15_14:26:16 INFO: IPaddr Success ResourceManager[1098]: 2007/03/15_14:26:16 info: Running /etc/init.d/serctl start heartbeat[1088]: 2007/03/15_14:26:17 info: local HA resource acquisition completed (standby). heartbeat[1064]: 2007/03/15_14:26:17 info: Standby resource acquisition done [foreign]. heartbeat[1064]: 2007/03/15_14:26:17 info: Initial resource acquisition complete (auto_failback) heartbeat[1064]: 2007/03/15_14:26:23 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:26:28 info: linux-xczz wants to go standby [foreign] heartbeat[1064]: 2007/03/15_14:26:28 info: standby: prueba2 can take our foreign resources heartbeat[1492]: 2007/03/15_14:26:28 info: give up foreign HA resources (standby). ResourceManager[1502]: 2007/03/15_14:26:28 info: Releasing resource group: prueba2 xxx.xxx.x.125/24 safe_asterisk ResourceManager[1502]: 2007/03/15_14:26:28 info: Running /etc/init.d/safe_asterisk stop ResourceManager[1502]: 2007/03/15_14:26:28 info: Running /usr/local/etc/ha.d/resource.d/IPaddr xxx.xxx.x.125/24 stop IPaddr[1561]: 2007/03/15_14:26:29 INFO: IPaddr Success heartbeat[1492]: 2007/03/15_14:26:29 info: foreign HA resource release completed (standby). heartbeat[1064]: 2007/03/15_14:26:29 info: Local standby process completed [foreign]. heartbeat[1064]: 2007/03/15_14:26:30 WARN: 1 lost packet(s) for [prueba2] [68:70] heartbeat[1064]: 2007/03/15_14:26:30 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:26:30 info: No pkts missing from prueba2! heartbeat[1064]: 2007/03/15_14:26:30 info: Other node completed standby takeover of foreign resources. heartbeat[1064]: 2007/03/15_14:27:00 info: prueba2 wants to go standby [foreign] heartbeat[1064]: 2007/03/15_14:27:11 info: standby: acquire [foreign] resources from prueba2 heartbeat[1784]: 2007/03/15_14:27:11 info: acquire local HA resources (standby). ResourceManager[1794]: 2007/03/15_14:27:11 info: Acquiring resource group: linux-xczz xxx.xxx.x.124/24 serctl IPaddr[1818]: 2007/03/15_14:27:11 INFO: IPaddr Running OK ResourceManager[1794]: 2007/03/15_14:27:11 info: Running /etc/init.d/serctl start ResourceManager[1794]: 2007/03/15_14:27:11 ERROR: Return code 1 from /etc/init.d/serctl ResourceManager[1794]: 2007/03/15_14:27:11 CRIT: Giving up resources due to failure of serctl ResourceManager[1794]: 2007/03/15_14:27:11 info: Releasing resource group: linux-xczz xxx.xxx.x.124/24 serctl ResourceManager[1794]: 2007/03/15_14:27:11 info: Running /etc/init.d/serctl stop ResourceManager[1794]: 2007/03/15_14:27:11 info: Running /usr/local/etc/ha.d/resource.d/IPaddr xxx.xxx.x.124/24 stop IPaddr[2090]: 2007/03/15_14:27:12 INFO: /sbin/route -n del -host xxx.xxx.x.124 IPaddr[2090]: 2007/03/15_14:27:12 INFO: /sbin/ifconfig eth0:0 xxx.xxx.x.124 down IPaddr[2090]: 2007/03/15_14:27:12 INFO: IP Address xxx.xxx.x.124 released IPaddr[2006]: 2007/03/15_14:27:12 INFO: IPaddr Success heartbeat[1784]: 2007/03/15_14:27:12 info: local HA resource acquisition completed (standby). heartbeat[1064]: 2007/03/15_14:27:12 info: Standby resource acquisition done [foreign]. heartbeat[1064]: 2007/03/15_14:27:12 info: remote resource transition completed. hb_standby[2228]: 2007/03/15_14:27:42 Going standby [foreign]. heartbeat[1064]: 2007/03/15_14:27:42 info: linux-xczz wants to go standby [foreign] heartbeat[1064]: 2007/03/15_14:27:42 info: standby: prueba2 can take our foreign resources heartbeat[2238]: 2007/03/15_14:27:42 info: give up foreign HA resources (standby). ResourceManager[2248]: 2007/03/15_14:27:43 info: Releasing resource group: prueba2 xxx.xxx.x.125/24 safe_asterisk ResourceManager[2248]: 2007/03/15_14:27:43 info: Running /etc/init.d/safe_asterisk stop ResourceManager[2248]: 2007/03/15_14:27:43 info: Running /usr/local/etc/ha.d/resource.d/IPaddr xxx.xxx.x.125/24 stop IPaddr[2310]: 2007/03/15_14:27:43 INFO: IPaddr Success heartbeat[2238]: 2007/03/15_14:27:43 info: foreign HA resource release completed (standby). heartbeat[1064]: 2007/03/15_14:27:43 info: Local standby process completed [foreign]. heartbeat[1064]: 2007/03/15_14:27:44 WARN: 1 lost packet(s) for [prueba2] [114:116] heartbeat[1064]: 2007/03/15_14:27:44 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:27:44 info: No pkts missing from prueba2! heartbeat[1064]: 2007/03/15_14:27:44 info: Other node completed standby takeover of foreign resources. What I am trying to have is ser running on linux-xczz and asterisk running on prueba2 with failover configured on both machines but apparently the failover crashes and I lose both my services if both heartbeats are running. Any idea why this happens or what I'm doing wrong?. Can ser and asterisk be run by heartbeat with failover support? thanxs in advance

2 1

[Serusers] Question about Radius Leg
by Gustavo Passos Tourinho 20 Mar '07

20 Mar '07

Hello I saw a Stop-Ack LEG and I would like to know what kind of signaling message could generate it. I mean, that 183 is Session Progress I know but why it generate Stop Ack leg? Shouldnt I have only start-ack (ACK send after 200 OK was received)? This is a quote of my radius log Thu Mar 15 13:49:18 2007 Acct-Status-Type = Stop Service-Type = IAPP-Register Sip-Response-Code = 183 Error-Cause = Ack Thanks, Gustavo

2 1

[Serusers] compiling the most recent unstable snapshots
by Eliott Spencer 20 Mar '07

20 Mar '07

I am having difficulty compiling the most recent unstable snapshots on Debian (Ubuntu Feisty Fawn) starting with the version released March 3rd. Previous versions compile well so I have been using the March 1st release and am learning a lot about the new SER (I really like @selects). My compile gets stuck at the eval module but when I skip that module, it gets stuck at the unixsock module. If I skip that module it gets stuck at another module and so on until when I finish compiling it won't start. Am I missing a key step or command? Thanks!

2 1

[Serusers] Problem loading mysql module
by POON Ben 20 Mar '07

20 Mar '07

Hi, I installed SER 0.9.3 for Solaris 10 (is there a newer version pre-compiled for solaris 10?). I'm using the sample config it comes with, except I uncommented the line: loadmodule "/usr/local/lib/ser/modules/mysql.so" Having done that, I get this error: 0(23162) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: ld.so.1: ser: fatal: libmysqlclient.so.10: open failed: No such file or directory I have installed MySQL already, and I checked that mysql.so is where it should be. Is there something else I have to configure to get MySQL running with SER? Thanks, Ben

4 3

[Serusers] Ser2 and radius - Accounting request doesn't send User-Name Attribute
by tzieleniewski 20 Mar '07

20 Mar '07

Hi, I am using radius for accounting and I have the following problem that whenever SER sends an Accounting message to Radius which contains the Acct-Status-Type = Start it doesn't send the User-Name attribute. Strange is that the User-Name is send with the massage with Acct-Status-Type = Stop. This disables radius to perform the correct accounting statistics because radius is unable to map the start and end of the session. Can it be somehow enabled in ser to send User-Name parameter in the first message? Below is an example of such two corresponding accounting messages: Mon Mar 12 14:46:17 2007 SER-Attr = "" Acct-Session-Id = "9e6a674c-0fcf-db11-94ea-00148539376e@sen" Sip-To-Tag = "36974326" SER-From = "\"Sen Server\" <sip:tzl@voip.touk.pl>;tag=406e674c-0fcf-db11-94ea-00148539376e" SER-Flags = 8 SER-Original-Request-ID = "sip:110@voip.touk.pl" Sip-Method = "INVITE" Sip-Cseq = 2 Sip-Translated-Request-ID = "sip:hellboy@192.168.0.105:7060" Sip-Source-IP-Address = 192.168.0.165 Sip-From-Tag = "406e674c-0fcf-db11-94ea-00148539376e" SER-To = "<sip:110@voip.touk.pl>;tag=36974326" SER-Digest-Username = "tzl" SER-Request-Timestamp = 1173707170 Calling-Station-Id = "sip:tzl@voip.touk.pl" Sip-Source-Port = 5067 SER-Digest-Realm = "voip.touk.pl" Sip-Response-Code = 200 Called-Station-Id = "sip:110@voip.touk.pl" SER-Response-Timestamp = 1173707177 Acct-Status-Type = Start Service-Type = IAPP-Register NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "16bd1ddc2a514fa0" Timestamp = 1173707177 Mon Mar 12 14:47:10 2007 SER-Attr = "" Acct-Session-Id = "9e6a674c-0fcf-db11-94ea-00148539376e@sen" Sip-To-Tag = "36974326" SER-From = "\"Sen Server\" <sip:tzl@voip.touk.pl>;tag=406e674c-0fcf-db11-94ea-00148539376e" SER-Flags = 8 SER-Original-Request-ID = "sip:hellboy@192.168.0.105:7060" Sip-Method = "BYE" Sip-Cseq = 4 Sip-Translated-Request-ID = "sip:hellboy@192.168.0.105:7060" Sip-Source-IP-Address = 192.168.0.165 Sip-From-Tag = "406e674c-0fcf-db11-94ea-00148539376e" SER-To = "<sip:110@voip.touk.pl>;tag=36974326" SER-Request-Timestamp = 1173707230 Calling-Station-Id = "sip:tzl@voip.touk.pl" Sip-Source-Port = 5067 Sip-Response-Code = 200 Called-Station-Id = "sip:110@voip.touk.pl" SER-Response-Timestamp = 1173707230 Acct-Status-Type = Stop Service-Type = IAPP-Register User-Name = "tzl(a)voip.touk.pl" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "4a9f1334616db4a4" Stripped-User-Name = "tzl" Realm = "voip.touk.pl" Timestamp = 1173707230

2 1

Re: [Serusers] SIP Accounting Module problem
by Greger V. Teigre 20 Mar '07

20 Mar '07

Copy serusers. Private emails that should go on SER end up in a different inbox and get lower priority... I'm not a solaris person, but ser logs to syslog. g-) POON Ben wrote: > Hi Greger, > > Thanks for the tip. I'm using Solaris 10 and whatever Ser version > it's bundled with, everything came preinstalled. I went to /var/log, > but there's no messages folder, all I have a syslog.# files. How can > I find out where the log messages are kept, if there are any at all? > Is that something I can set in the ser.cfg file? > > Thanks, > Ben > > ------------------------------------------------------------------------ > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* Friday, March 09, 2007 1:56 AM > *To:* POON Ben > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] SIP Accounting Module problem > > Ben, > the cat error just tells you that ser didn't start. Find the command > in your ser service script that actually starts ser and start it > manually. You may also add -E -D -D -D -D -D > to get lots of debug info to stderr (or just look in your > /var/log/messages) > g-) > > POON Ben wrote: >> Hi folks, >> >> I'm trying to get my accounting services working, but am having >> trouble doing so... I am using the sample ser.cfg file, and am >> having a problem loading the acc.so module. >> >> In the ser.cfg file, I try added the line: loadmodule >> "/usr/sfw/lib/ser/modules/acc.so" to the bottom of the loadmodule >> section. I haven't done any modparam or scripting yet, even so, when >> I restart the ser service, I get: Starting SER : cat: cannot open >> /var/run/ser.pid. >> >> If I comment or remove the loadmodule line, ser will load again. Is >> there anything else I need to add or configure before loading this >> module? I went to /usr/sfw/lib/ser/modules to make sure acc.so is >> there, and it is. >> >> Any help is greatly appreciated. Thank you very much. >> Ben >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >>

1 0

[Serusers] Change URI before accounting ?
by inge 20 Mar '07

20 Mar '07

Hi all, When I modify an URI by stripping some words, in ACC table I've got the old URI, before the stripping. So if INVITE contains sip:7000123456789, after strip he contains sip:0123456789, in acc I've INVITE ACK and BYE with sip:7000123456789. How can I solve it ? Thanks for your help. Regards, Adrien

3 11

[Serusers] SER home proxy with Asterisk voicemail
by Meidiana Ten 20 Mar '07

20 Mar '07

Hi guys, I'm give up with SEMS as media server for voicemail cause it can't successly run although I've followed how to install SEMS at http://ftp.iptel.org/pub/sems/doc/current/Configure-Sems-Ser-HOWTO.html and there is no answer for the problem So I decide to try Asterisk as the voicemail server. But I never use Asterisk before. I really need help in configuring Asterisk. What should I have (the softwares) before compiling and configuring Asterisk as voicemail server. Thanx for the attention. Regards, Meidiana ____________________________________________________________________________________ Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html

5 9

[Serusers] Call forwarding
by Jai Rangi 20 Mar '07

20 Mar '07

Hello, I have a ser and asterisk setup. Users register on ser and all the national and international calls go through asterisk. Asterisk handles the billing part for long distance calls. Now I got an issue with call forwarding. Here is the situation. Useragent A calls Usergent B. User B has setup the call the call forwarding to his cell phone. So when user A calls user B, ser check that user B has set the forwarding and send the call to Asterisk box. Asterisk box sees user A in the callerID and bills user A for that call. We want user B to be billed for that call, cause he is forwarding the calls to his cell phone (long distance number). How can I achieve this. Thank you, -Jai

1 0

[Users] permissions module and avp definition
by Andreas Granig 19 Mar '07

19 Mar '07

Hi, It seems like most (all?) of the module params defining an AVP have changed from just the name (like "i:100") to a full avp spec (like "$avp(i:100)"). Does this also hold true for peer_tag_avp in permissions module? I ask because it's still "i:707" in the example of http://openser.org/docs/modules/1.2.x/permissions.html#AEN343 and it's not mentioned in http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x Regards, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

2 1

[Users] t_uac_dlg disconnect example - WILLING TO PAY
by Daryl Sanders 19 Mar '07

19 Mar '07

I need a complete example (with proper syntax) of how to disconnect a call (send BYE) using "t_uac_dlg". This can be either via bash script or openserctl command. I have been fighting with this for some time and always seem to get "Too few or to many parameters" errors. I have been through the ctd.sh example and searched the mailing list but, I'm still having no luck. I have read Jan's "SER FIFI Interface" doc, and I believe my fifo files are working properly, however I really don't know the exact syntax, or what elements are required to get this to work. I have access to the "Call-ID" as it is stored in MySQL via acc. I am willing to pay for this example. Just let me know a fair price. FYI: I am using OpenSER 1.2. Thanks, Daryl

2 1

[Users] Permissions module w/DBTEXT database doesn't work in 1.2 ???
by Daryl Sanders 19 Mar '07

19 Mar '07

Hi Everyone, I cannot seem to get the permissions module with dbtext to load in version 1.2. I am pretty sure the "trusted" table format is correct. This is what I have. src_ip(str) proto(str) from_pattern(str) tag(str,null) 192.168.0.10:any:^sip\:.*$: I get the following error when I start OpenSER... ERROR: permissions: reload_address_table(): Error while querying database Regards, Daryl

3 5

[Users] using avp's as lists
by Kro Kamil 19 Mar '07

19 Mar '07

Hi, can i use avp's as a list holder? what i want to do is using avp_db_query function to get the list of routes which will be used for forwarding calls. for example avp_db_query("select route_ip from tbl_route","$avp(s:route_list)"); then rewritehost("$avp(s:route_list)[1]"); on failure rewritehost("$avp(s:route_list)[2]"); thanks in advance _________________________________________________________________ Live.com'u deneyin: çevrimiçi dünyanızı bir araya getirin; haberler, spor, hava durumu ve çok daha fazlası. http://www.live.com/getstarted

2 1

[Users] Manipulate Refer To
by Sam Lee 19 Mar '07

19 Mar '07

Hey guys, How do you manipulate the 'Refer to' header in the REFER Method ? I need to do some string manipulation to change it to something that i want. Thanks ahead! Regards, Sam

3 8

[Users] Server call forwarding
by Sam Lee 19 Mar '07

19 Mar '07

Hey guys, I'm thinking of try out a server side forwarding where forwarded number is stored in a database. I just want to ask if anyone has done it before and how was the call routing done . Been thinking alot but couldn't figure out a way to do it. Any pointers ? Regards, Sam

3 2

[Users] WeSIP session question
by tele 19 Mar '07

19 Mar '07

Hi, I've a problem with WeSIP in B2BUA mode, in case of failed call 4xx-6xx correctly terminated, when i try to send another call to WeSIP i recive a "473 Filtered destination" then for send another call i've to wait WeSIP complete some management with session: 14:10:16 19Mar2007 DEBUG SipConnector [SipProcessor[3]]- recycle: Recycling processor SipProcessor[3] 14:10:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 14:10:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- SipSession [z9hG4bK69e6006f] in state [3] with lifetime of :74490 14:10:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 14:10:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- SipSession [z9hG4bK69e68616] in state [3] with lifetime of :40281 14:11:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 14:11:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- SipSession [z9hG4bK69e6006f] in state [3] with lifetime of :134500 14:11:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 14:11:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- SipSession [z9hG4bK69e68616] in state [3] with lifetime of :100291 14:12:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =0 14:12:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- Remove AppSession [B4D6C9C4288784A68E032D20AB78BD8E] When i see Remove AppSession i'm able to send another call... I've read the sip servlet spec about that and i'm trying to invalidate() or setExpires() to SiApplicationSession in case of failed call. but it's not clear how to do yet. i can provide the full debug if needed. thank you! regards :tele

1 0

[Users] WeSIP with OpenSER 1.2.0
by tele 19 Mar '07

19 Mar '07

Hi, I've upgraded openser from 1.2.0-pre6 to 1.2.0 with the seas module loaded. try to send a call and this is the output: 5(24664) SIP Request: 5(24664) method: <INVITE> 5(24664) uri: <sip:390104491079@82.215.163.67> 5(24664) version: <SIP/2.0> 5(24664) parse_headers: flags=2 5(24664) Found param type 232, <branch> = <z9hG4bK5d297af7>; state=16 5(24664) end of header reached, state=5 5(24664) parse_headers: Via found, flags=2 5(24664) parse_headers: this is the first via 5(24664) After parse_msg... 5(24664) preparing to run routing scripts... 5(24664) parse_headers: flags=100 5(24664) DEBUG:maxfwd:is_maxfwd_present: value = 70 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) DEBUG:parse_to:end of header reached, state=9 5(24664) DBUG:parse_to: display={}, ruri={sip:390104491079@82.215.163.67} 5(24664) DEBUG: get_hdr_field: <To> [32]; uri=[sip:390104491079@82.215.163.67] 5(24664) DEBUG: to body [sip:390104491079@82.215.163.67 ] 5(24664) get_hdr_field: cseq <CSeq>: <1> <INVITE> 5(24664) DEBUG: get_hdr_body : content_length=213 5(24664) found end of header 5(24664) DEBUG: t_newtran: T on entrance=0xffffffff 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) parse_headers: flags=78 5(24664) t_lookup_request: start searching: hash=4873, isACK=0 5(24664) DEBUG: RFC3261 transaction matching failed 5(24664) DEBUG: t_lookup_request: no transaction found 5(24664) DEBUG: add_param: tag=5d297af7 5(24664) DEBUG:parse_to:end of header reached, state=29 5(24664) DBUG:parse_to: display={}, ruri={sip:3405300695@82.215.163.5} 5(24664) new INVITE 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) check_via_address(82.215.163.5, 82.215.163.5, 0) 5(24664) WARNING:vqm_resize: resize(0) called 5(24664) DEBUG:tm:_reply_light: reply sent out. buf=0x814fa58: SIP/2.0 1..., shmem=0xb60a0680: SIP/2.0 1 5(24664) DEBUG:tm:_reply_light: finished 5(24664) as found ! (app_server_one) processor id = 1 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) msglen = 561,msg starts at 167 5(24664) DEBUG:tm:UNREF_UNSAFE: after is 0 5(24664) DEBUG:destroy_avp_list: destroying list (nil) 5(24664) receive_msg: cleaning up -1(0) Event relaied to app_server_one AS -1(0) Event type FINAL_REPLY -1(0) Scanning data from 1 AS -1(0) Scanning data from 0 uncomplete AS -1(0) polling [2 ServSock] [1 pipe] [1 App Servers] [0 Uncomplete AS] java.lang.StringIndexOutOfBoundsException: String index out of range: -19 at java.lang.String.checkBounds(String.java:368) at java.lang.String.<init>(String.java:404) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:30) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPRequest.<init>(SIPRequest.java:154) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:355) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:135) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:141) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595)

2 2

[Users] Failure routes
by Sam Lee 19 Mar '07

19 Mar '07

Hey guys , I noticed that only 1 failure routes is valid for a t_relay . Is that true or i've done wrong with my config. What i did was : If condition A { t_on_failure("1") } If condition B { T_on_failure("2") } t_relay If it can't be done , how am i suppose to do condition A or B ? Regards, Sam

2 3

[Users] error adding a new subscriber
by rkmr.em＠gmail.com 19 Mar '07

19 Mar '07

Hi I am adding a subscriber and I get this error below: /usr/local/sbin/openserctl add 100 100 100@localhost database engine 'PGSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter= ERROR: user '100' already exists as alias I am sure this user does not exist as I checked the table subscriber and there is only one row with user 'admin' Thanks mark

2 4

[Users] RTPProxy problem with music-on-hold
by Federico Giannici 18 Mar '07

18 Mar '07

I have a problem with RTPProxy: often when I put a call on hold the callee hear no music. No problem if I don't pass through RTPPropxy. The strangest thing is that, if I do a lot of tries, I eventually succeed to make the callee to hear the music! I put rtpproxy in debug mode and found that when I finally succeed the usual line: callee's address filled in: 83.211.227.15:63829 (RTCP) is replaced by: callee's address filled in: 83.211.227.15:63828 (RTP) guessing RTCP port for callee to be 63829 I don't know RTP enough to understand if all these is normal or some kind of bug. Somebody have a clue? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

1 0

[Users] openser + postgres starting problem
by rkmr.em＠gmail.com 18 Mar '07

18 Mar '07

Hi If I compiled openser with postgres module and I get the following errors: [mark@localhost|~|16:34:34] /usr/local/sbin/openserctl moni /usr/local/lib/openser/openserctl/openserctl.pgsql: line 57: unexpected EOF while looking for matching `'' /usr/local/lib/openser/openserctl/openserctl.pgsql: line 70: syntax error: unexpected end of file database engine 'PGSQL' loaded Control engine 'FIFO' loaded ERROR: Error opening OpenSER's FIFO FIFO ERROR: Make sure you have line 'fifo=FIFO' in your config The line 57 3rd line in here: # input: sql query, optional pgsql command-line params pgsql_ro_query() { mdbg "pgsql_ro_query: $PGSQL $2 -h $DBHOST -U $DBROUSER $DBNAME -c '$1'" PGPASSWORD="$DBROPW" $PGSQL $2 \ -h $DBHOST \ -U $DBROUSER \ $DBNAME \ -c "$1" } How to fix this? thanks

3 9

Re: Fw: [Users] TM : retransmission timers
by Rao Ramaratnamma 18 Mar '07

18 Mar '07

Hi Weiter, Yeah, I have been trying to limit myself to technical observations too, but the governance aspect is somewhat interesting too as a hint for future development, even though I guess even this is much more confusing than the technical ones. I have investigated, both projects have their firms with them that pursue their commercial interests which creates a risk of possibly departing from the public interest, like with redhat. From this angle they look quite similar. But if any worries me just a little bit more than openser. Appearance at commercial shows on the "open" side versus technical event on the "net" side if I take your BSD parallel, marketing "open" webpage accusing "net" version bad, hiding root commerical sponsors on the "open" webpage, this could be signs for a redhat-like doubleedged sword. Hopefully I am oversensing because I mean it is natural that everybody has SOME interest, but indisputably folks on both sides have done good work, but same indisputably more TRANSPARENCY would be helpful for both projects so that users can be less investigative. But I agree the technical comparison you suggest will be very useful if not most useful. This is what I am eventually upto. Anything folks have to tell in this topic is most welcome like the retransmission timers in subject or user loading. rr disconcerted by the fact that the more I know the more I am confused and determined to get over the learning curve quickly. also excuse the abuse I crossposted again but I think cross interrogation is a bit painful but the more effective :-) ----- Original Message ---- From: Weiter Leiter <bp4mls(a)googlemail.com> To: Kim Il <kim_il_s(a)yahoo.com> Cc: users(a)openser.org Sent: Thursday, November 9, 2006 1:42:29 PM Subject: Re: Fw: [Users] TM : retransmission timers Common user barely has time to meet his boss requirements, rather than playing around with different scenarios, platforms, environments. I only read one email where Daniel stated that OpenSER now performs a whole much better while loading users from database. SER guys put no figure out yet, neither bare numbers nor comparisons. I'm just really curious to see how both servers perform, that's all. Even though I must maintain my SER, I kinda like OpenSER's faster releases and developers' responsiveness (that I shamelessly exploit for the common code left there :-), which is pretty much nonexistent with iptel (at least this is the general belief here at OpenSER). But about this I'll probably have to fight on SER's mailing list. I still wish that one day I won't have to compare features; heck, NetSER and FreeSER are still available ;-). WL. PS. Maybe regretfully, I haven't seen any iptel booth at von this year, while OpenSER guys put up a nice show. My congrats. On 11/9/06, Kim Il <kim_il_s(a)yahoo.com> wrote: I can see what you are hinting at, but I guess that the users are the unbiased party that should do the judgment and not the parties who have something to gain. cheers Weiter Leiter <bp4mls(a)googlemail.com> wrote: This features comparisons are not to last for too long, some performance comparisons would also be nice. After all, there are plenty of UA-level stacks out there. At least now that both projects get to have stable releases after forking and some core functionality remained shared. I wonder what "unbiased" organization will take up the challenge. :-) On 11/8/06, Kim Il < kim_il_s(a)yahoo.com > wrote:Mike, this is a really good start and we should collect these things so as to help the community to take the right choice. I would also suggest that what ever ground breaking issues we list we stay at the functional level (I do not think anyone is helped by using a description containing "allowing carrier grade platforms" and similar marketing phrases). cheers {truncated because too large} Sponsored Link Talk more and pay less. Vonage can save you up to $300 a year on your phone bill. Sign up now. _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

19 63

[Users] Openser architectural question
by vallimamod abdullah 17 Mar '07

17 Mar '07

Hello List, I have currently one openser proxy running on an Intel Xeon 2.8GHz and which is offering different sip customer services. It works fine, the only problem is the complexity of the configuration. For better security and easier maintenance, I have imagined to have one openser instance per service running on a virtual server (linux vserver). I would like to hear from the community opinion about the kind of problems I may encounter with this schema (performance limitations over all). If somebody have already built such architecture, I'll be happy to learn from his experience ! Thanks. -- Regards, -vma .

2 1

[Serusers] SER doesn't load but system thinks it is
by POON Ben 16 Mar '07

16 Mar '07

Hi all, I am trying to load ser, and it says it does. But when I do a ps -el | grep ser, it doesn't show up... I am running Solaris 10, ser 0.9.3. Can anyone please give me a hand? This is what I did: ************************************************************************ ****************** bash # ./serctl start Starting SER : started pid(532) bash # ps -el | grep ser 0 S 0 421 402 0 40 20 ? 695 ? ? 0:00 htt_serv 0 S 0 384 1 0 40 20 ? 253 ? ? 0:00 dpkeyser 0 S 0 390 1 0 40 20 ? 232 ? ? 0:00 jserver 0 S 0 391 390 0 40 20 ? 328 ? ? 0:00 jserver_ bash # ./serctl restart Stopping SER : ./serctl: line 812: kill: (532) - No such process stopped Starting SER : PID file exists! (/var/run/ser.pid) already running? ************************************************************************ ****************** The system thinks it's already running, but there's no actual process started... Any help is greatly appreciated. Thank you very much, Ben

2 1

[Users] OT: Troubleshooting DNS SRV records
by kjcsb 16 Mar '07

16 Mar '07

I'm trying to get SRV records working and am having difficulties. I wonder if anyone could point me in the direction of some useful information on this topic to assist in troubleshooting. I've read a lot of stuff about how to set it up and check the records, but since I'm still having problems, there's something else wrong and I've run out of ideas. nslookup set type=srv _sip._tcp.domain.co.nz Server: 64.38.5.242 Address: 64.38.5.242#53 Non-authoritative answer: _sip._tcp.domain.co.nz service = 10 10 5060 domain.co.nz. _sip._tcp.domain.co.nz service = 20 10 5060 beta.domain.co.nz. Authoritative answers can be found from: domain.co.nz nameserver = ns1.widge.net. domain.co.nz nameserver = ns2.widge.net. > _sip._udp.domain.co.nz Server: 64.38.5.242 Address: 64.38.5.242#53 Non-authoritative answer: _sip._udp.domain.co.nz service = 10 10 5060 domain.co.nz. _sip._udp.domain.co.nz service = 20 10 5060 beta.domain.co.nz. Authoritative answers can be found from: domain.co.nz nameserver = ns2.widge.net. domain.co.nz nameserver = ns1.widge.net. One thing I notice from this is that no IP addresses are returned. Is that normal? The problems I am experiencing are that neither my Grandstream BT102 or SPA3000 can resolve the proxy name using SRV (when use SRV is set to on in those devices). In the case of the BT102 it won't register (switching off use SRV allows it to register). In the case of the SPA3000 when I view the syslog output I can see it failing over from the SRV record to the A record which then resolves. Any help appreciated. Cameron

2 1

[Users] Re: help in uac config
by vallimamod abdullah 16 Mar '07

16 Mar '07

On 3/16/07, Jobson Andrade <jandrade(a)obelisknet.com.br> wrote: > I made this configuration but not work, i need send International calls for > my gateway using one identification (username and password) only this more > not work my configuration. I though you just wanted to change the From header. The 'uac_replace_from()' just changes this header in the sip message. To make client authentification, you need to configure a failure route and use 'uac_auth()' in it. There is an example showing exactly what you are looking for at: http://www.voice-system.ro/docs/uac/ar01s06.html#ex_auth -- Regards, -vma .

1 0

[Users] Re: help in uac config
by vallimamod abdullah 16 Mar '07

16 Mar '07

Hello Jobson, On 3/16/07, "Jobson Andrade" <jandrade(a)obelisknet.com.br> wrote: > 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? IMHO, it shoud be, according to your example: uac_replace_from(" sip:username@itsp.com.br"); Here are 2 pointers to interesting documentation and example: http://www.openser.org/docs/modules/1.2.x/uac.html#AEN174 http://www.voice-system.ro/docs/uac/ar01s06.html Hope this will help. -- Regards, -vma .

2 1

[Users] Openser = 1.1.1 init_tls: null openssl compression methods
by Bodin Bruno 16 Mar '07

16 Mar '07

I use Fedora Core 5 and when I launch openser in TLS mode : 0(13701) init_tls: Entered 0(13701) WARNING:init_tls: disabling compression due ZLIB problems 0(13701) ERRRO:init_tls: null openssl compression methods 0(13701) could not initialize tls, exiting... How can I fix this ? Thank :)

1 0

R: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem
by Fabio Macchi 16 Mar '07

16 Mar '07

Hello, I'm using Nathelper + RTPproxy ( can't use Nathelper-only because my gateway doesn't support passive mode ) : route part of ser.cfg attached. In the invite block I use force_rport before authorize block ( as previous discussed ) and the NAT block after: route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } The original BYE comes from Gateway with the wrong request URI ( private UA ip ) Request-Line: BYE sip:xxxxxx@1.255.17.9 SIP/2.0 Method: BYE [Resent Packet: False] Fabio -----Messaggio originale----- Da: Kostas Marneris [mailto:K.Marneris@otenet.gr] Inviato: venerdì 16 marzo 2007 13.19 A: Fabio Macchi Cc: Kostas Marneris Oggetto: Re: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hello, Which 'NAT Traversal mechanism' do you use ? Nathelper only or Mediaproxy ? Do you change the Contact address somewhere in INVITE block ? force_rport(); fix_nated_contact(); fix_nated_sdp("3"); Check (with a tcpdump at SER) how the original BYE comes from GW. What is the R-URI at the original BYE request (comes from GW) ? Kostas Fabio Macchi wrote: > First, thanks for answer. > > I've tryed your trik and in effect this solve the problem of the '200 ok' > forwarded to the UA, but my problem still remain alive: when BYE is sent > from Gateway, it reaches correctly SER, but it still forward it to the > private UA address. I was wondering about the nat_uac_test in this case, as > the source of the BYE message is the gateway ( not natted ) and not the UA. > > Have any idea about this ? > > Fabio > > -----Messaggio originale----- > Da: Kostas Marneris [mailto:K.Marneris@otenet.gr] > Inviato: giovedì 15 marzo 2007 20.39 > A: Fabio Macchi > Cc: serusers(a)lists.iptel.org > Oggetto: Re: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem > > Hello, > I was working on about the same problem today either with 'Mediaproxy > solution' > or with 'SER's Nathelper only solution' . > > The NAT issue is a nightmare, not because of SER but because of > different implementations on NAT boxes. > > Actually my problem was : > if the NATed UA send a BYE to SER, SER forward it to PSTN-GW, > then the '200 Ok' Response from PSTN-GW is forwarded by SER to UA > to the wrong port (Contact or Via header port). > > I used the following block on Loose Route section, > (because BYE is loose_routed if you use Record-Route), > and it seems to work. > > # --------------------------------------- > # Loose Route Section > # --------------------------------------- > if (loose_route()) { > # mark routing logic in request > if (method == "BYE") { > if (nat_uac_test("22")) { > xlog("L_NOTICE", "*** LR -> NATed BYE - Use > force_rport()"); > force_rport(); > }; > }; > route(1); > break; > }; > > > > > I faced up your second problem too. > The solution was to move the NAT handling block before proxy_authorize > block. > > I think that the different behaviour does not come with the 'standard > RFC1918 addresses', > but with the different NAT type. > > I realize that the provisional mesgs '100 Trying' and '407 Proxy > Authentication Required' > are relayed back to the real IP addr of NATed UA (this is correct), > but to the WRONG port (that of Contact/Via header and not the signalling > received port). > It seems that these mesgs use the IP address part of 'Received' field of > Location DB > but not the port. > > It happens to work if NAT box use the SAME port (eg. 5060) on NAT > translation > (10.10.10.1:5060 --> Real_IP:5060) (eg. with a SAGEM1500 Router) > But it does not work if NAT box doesn't use the same port > (10.10.10.1:5060 --> Real_IP:38181) > > > I think that this has to be verified by SER developers or SER experts. > > > > Kostas > > --- > K.Marneris(a)otenet.gr > > ----- Original Message ----- > From: "Fabio Macchi" <f.macchi(a)keeptelecom.com> > To: <serusers(a)lists.iptel.org> > Sent: 15 March 2007 19:34 > Subject: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem > > >> Hi all, >> >> >> >> I'm running the following schema: >> >> >> >> UA ( possibly natted ) -> SER -> PSTN Gateway >> >> >> >> I have a problem with UA belonging to a particular network with private >> address not RFC1918 compliant ( class 1.x.x.x ), SER and PSTN Gateway have >> pubblic address. >> >> >> >> The problem is that, after a succesfull call, if the PSTN gateway send a > BYE >> to SER, then SER forward BYE to the private address of UA instead of > pubblic >> one. >> >> >> >> I don't understand which is the section that handle BYE messages and how > can >> I solve this problem: anyone may help ? >> >> >> >> >> >> >> >> Second, another question: with this particular network I had problem with >> INVITE too, because SER was sending "proxy authorization request" to the >> wrong TCP port. To solve this, I've moved the nat handling ( with >> force_rport ) before the proxy_authorize block and it's working, but why >> this is not necessary on standard RFC1918 compliant natted address ? >> >> >> >> Many thanks for any explanation >> >> >> >> Fabio >> >> > > > > -- Kostas Marneris e-mail: K.Marneris(a)otenet.gr Tel : +30-210-6151886

1 0

[Users] Problem with dialog module
by Leonid Fainshtein 16 Mar '07

16 Mar '07

Hi, I see that when Openser 1.1 receives re-INVITE it removes parameter "did=" from header field Route before sending it to the second call party. Any idea? Thank you in advance, Leonid ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

2 1

[Users] AVPOPS: question about avp_db_query
by Papadopoulos Georgios 16 Mar '07

16 Mar '07

Hello, Is there a way to load an integer value into an integer avp using avp_db_query? For example if in table usr_preferences I have value='15' and type=1, then using avp_db_load I get an avp with an int value of 15. Then I can use this int avp to check whether it is greater or less than some value. I am trying to do the same thing with avp_db_query but the avp that I get is a string. 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.

3 3

[Users] TLS between OPENSer and FreeRadius ?
by Bodin Bruno 16 Mar '07

16 Mar '07

Hello, Is it possible to use a TLS tunnel between OpenSer and Radius ? thank

1 0

[Serusers] make all error: avp_radius module compilation error - parse_uri.h:50
by tzieleniewski 16 Mar '07

16 Mar '07

Hi, I have just downloaded the cvs version of ser 2.1.0-dev1 and after I run #make group_include="radius" all (like usually) I got the following error: make[1]: Entering the directory `/home/ser/src/sip_router/modules/avp_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev1"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -I/usr/local/include -c avp_radius.c -o avp_radius.o avp_radius.c:244: error: static declaration of ‘uri_type_to_str’ follows non-static declaration ../../parser/parse_uri.h:50: error: previous declaration of ‘uri_type_to_str’ was here make[1]: *** [avp_radius.o] Error 1 make[1]: Opuszczenie katalogu `/home/ser/src/sip_router/modules/avp_radius' make: *** [modules] Error 1 Cheers Tomasz

2 1

[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

3 5

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 >

2 1

[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

1 0

[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

1 0

[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

2 1

[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

2 2

[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 */

2 1

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users