sr-users March 2007

sr-users@lists.kamailio.org

192 participants
381 discussions

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

07 Mar '07

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

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

07 Mar '07

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

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

07 Mar '07

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

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

07 Mar '07

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

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

06 Mar '07

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

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

06 Mar '07

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

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

06 Mar '07

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

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

06 Mar '07

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

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

06 Mar '07

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

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

06 Mar '07

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

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sr-users March 2007