Re: [OpenSIPS-Users] OpenSIPS 1.8 segfault
Hello, Can you please provide the OpenSIPS full debug log as well as the full SIP trace for your scenario that causes the crash ? You can send these privately to me if they contain sensitive info. Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 05/19/2012 02:33 AM, Daniel Goepp wrote: Just testing out a new lab box preparing for upgrading our network to 1.8, and we get a segfault right away. Not all call produce this, but most from my desk phone seem to. I can reproduce this pretty easily, any thoughts? Or more data that would help investigate this? #0 0x004f3913 in fm_remove_free (n=0x7fc347c5a700, qm=0x7fc347be2010) at mem/f_malloc.c:172 #1 fm_malloc (qm=0x7fc347be2010, size=8) at mem/f_malloc.c:378 #2 0x00461139 in hostent_cpy (dst=0x7fc347c5b620, src=0x7fc348399de0) at proxy.c:163 #3 0x0046219f in mk_proxy (name=0x7fff45288300, port=optimized out, proto=optimized out, is_sips=0) at proxy.c:258 #4 0x7fc346ac0307 in uri2proxy (uri=0x7fc346d0c7e0, forced_proto=0) at ut.h:111 #5 0x7fc346ac14bf in add_uac (t=0x7fc33f9f64c0, request=0x7fc346d0c580, uri=optimized out, next_hop=optimized out, path=optimized out, proxy=optimized out) at t_fwd.c:410 #6 0x7fc346ac4c1f in t_forward_nonack (t=0x7fc33f9f64c0, p_msg=0x7fc346d0c580, proxy=0x0) at t_fwd.c:658 #7 0x7fc346ad49c0 in w_t_relay (p_msg=0x7fc346d0c580, proxy=0x0, flags=optimized out) at tm.c:1152 #8 0x004194b1 in do_action (a=0x7fc347c13d38, msg=0x7fc346d0c580) at action.c:1478 #9 0x0041ddb9 in run_action_list (a=optimized out, msg=0x7fc346d0c580) at action.c:143 #10 0x0041b254 in run_actions (msg=0x7fc346d0c580, a=optimized out) at action.c:123 #11 run_actions (msg=0x7fc346d0c580, a=optimized out) at action.c:281 #12 do_action (a=0x7fc347c10420, msg=0x7fc346d0c580) at action.c:568 #13 0x0041ddb9 in run_action_list (a=optimized out, msg=0x7fc346d0c580) at action.c:143 #14 0x0041bad4 in do_action (a=0x7fc347c104f8, msg=0x7fc346d0c580) at action.c:911 #15 0x0041ddb9 in run_action_list (a=optimized out, msg=0x7fc346d0c580) at action.c:143 #16 0x0041e140 in run_actions (msg=0x7fc346d0c580, a=0x7fc347c0d290) at action.c:123 #17 run_actions (msg=0x7fc346d0c580, a=0x7fc347c0d290) at action.c:168 #18 run_top_route (a=0x7fc347c0d290, msg=0x7fc346d0c580) at action.c:184 #19 0x7fc346ae4e31 in run_failure_handlers (t=0x7fc33f9f64c0) at t_reply.c:651 #20 t_should_relay_response (Trans=0x7fc33f9f64c0, new_code=optimized out, branch=0, should_store=0x7fff45289b90, should_relay=optimized out, cancel_bitmap=optimized out, reply=0x7fc347c591c8) at t_reply.c:950 #21 0x7fc346ae4ec8 in relay_reply (t=0x7fc33f9f64c0, p_msg=0x7fc347c591c8, branch=0, msg_status=302, cancel_bitmap=0x7fff45289c88) at t_reply.c:1164 #22 0x7fc346ae78a9 in reply_received (p_msg=0x7fc347c591c8) at t_reply.c:1539 #23 0x0043458b in forward_reply (msg=0x7fc347c591c8) at forward.c:574 #24 0x00480b12 in receive_msg ( buf=0x7fc33f9f95f8 SIP/2.0 302 Moved temporarily\r\nVia: SIP/2.0/TCP x.x.x.x;branch=z9hG4bK0af7.85b09c26.0;i=4\r\nVia: SIP/2.0/TCP x.x.x.x:5060;received=x.x.x.x;branch=z9hG4bK03b2f52e6a4ffa0e350aad1e51c88e26.1;r..., len=optimized out, rcv_info=0x7fc33f9f9588) at receive.c:203 #25 0x004c6255 in tcp_read_req (con=0x7fc33f9f9578, bytes_read=0x7fff45289ec0) at tcp_read.c:546 #26 0x004c6bdc in handle_io (fm=0x7fc347c4ad50, idx=-1) at tcp_read.c:817 #27 0x004c97bb in io_wait_loop_epoll (repeat=optimized out, h=optimized out, t=optimized out) at io_wait.h:728 #28 tcp_receive_loop (unix_sock=optimized out) at tcp_read.c:937 #29 0x004c44be in tcp_init_children (chd_rank=0x7fcd00) at tcp_main.c:1819 #30 0x00416c47 in main_loop () at main.c:941 #31 main (argc=optimized out, argv=optimized out) at main.c:1520 -dg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Source IP address on Asterisk integration
I guess I'm wondering why you want to do that? Mark On Mon, May 21, 2012 at 5:49 PM, ksy ksybl...@gmail.com wrote: Ronald Cepres rbcepres@... writes: Hi all, I'm trying to set up Opensips so that it simply relays the requests it receives to Asterisk on the same server, only using a different port. The set-up is working but my problem is Asterisk uses the IP of OpenSIPS as peer contact even if the domain on the Contact header is from the actual sender of the request. The peer's setting on Asterisk is nat=yes, and I'm not allowed to change this value. Can I tweak something on Opensips so that Asterisk can see the real sender's IP address even with nat=yes on Asterisk? Thanks! Regards, Ronald Hi, Ronald! Have the same issue. I wonder if you have solved the problem. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS Control Panel Release 5.0
Hi all, We have a new release for the OpenSIPS Control Panel: 5.0. This new release comes with a heavily modified Dynamic routing module in order to be compatible with OpenSIPS v.1.8.x The only major changes are related to the Dynamic routing module (there are also some bug fixes). Best regards, Alex Ionescu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Advance SBC
Hi, I am looking on wiki about implementing SBC with mod_easyroute and mod_lcr (http://wiki.freeswitch.org/wiki/Advance_SBC_with_mod_lcr_and_mod_easyroute). I have configured as is written on wiki in everything works. Just one question. Should mod_lcr automatically be routing to less cost destination if first destination is down (this does not happend)? Regards, Miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Advance SBC
On 5/21/2012 12:44 PM, Miha wrote: Hi, I am looking on wiki about implementing SBC with mod_easyroute and mod_lcr (http://wiki.freeswitch.org/wiki/Advance_SBC_with_mod_lcr_and_mod_easyroute). I have configured as is written on wiki in everything works. Just one question. Should mod_lcr automatically be routing to less cost destination if first destination is down (this does not happend)? Regards, Miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users Sorry wrong destination email. Ignore this! Regards, Miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog vars not being sync'ed from DB to memory
Hello all, Is there any update on my question? Should I open a bug about this issue, or is there any other test I can run to verify this feature? Thanks, Mariana. On Thu, May 17, 2012 at 2:22 PM, Mariana Arduini marianardu...@gmail.comwrote: Hello, The dlg_db_sync command is only useful when you have the second server online, and want to trigger a refresh of OpenSIPS memory based on what is in the DB. In fact I noticed it. I also tried this test with no success in fetching the dialog vars: 1) Server #2 is online listening on 10.0.0.1 2) interface 10.0.0.1 is set down on server #2, but server #2 is not stopped 3) interface 10.0.0.1 is set up on server #1 and server #1 is started 4) UAC sends INVITE to 10.0.0.1, which goes to server #1 5) interface 10.0.0.1 is set down on server #1, and server #1 is stopped 6) interface 10.0.0.1 is set up on server #2 7) dlg_db_sync is run on server #2, but dlg_list_ctx shows no vars 8) UAC sends BYE to 10.0.0.1, which goes to server #2 If you just start the secondary server do not issue dlg_db_sync, do you still have the same problem ? Yes, no dialogs vars in context. Attached is the log on server #2 you asked for. It includes the following, in this order: 1) opensipsctl start 2) opensipsctl fifo dlg_list_ctx: no vars shown in context 3) opensipsctl fifo dlg_db_sync 4) opensipsctl fifo dlg_list_ctx: no vars shown in context 5) BYE from UAC, server looks for dialog var caller_tag For this test, I tried to get the vars using fetch_dlg_value(), file name is log-server-2-from-start-fetch.txt. I collected another log for the same test, now using get_dialog_info() instead of fetch_dlg_value(), file name is. Thanks for the help. Mariana On Thu, May 17, 2012 at 6:08 AM, Vlad Paiu vladp...@opensips.org wrote: Hello, Just to clear some things up, if you leave the second server offline and only start it after the active is down, then the ongoing dialogs will be automatically loaded by the secondary server at startup. The dlg_db_sync command is only useful when you have the second server online, and want to trigger a refresh of OpenSIPS memory based on what is in the DB. If you just start the secondary server do not issue dlg_db_sync, do you still have the same problem ? If you can, please send us ( privately or via pastebin ) a full debug OpenSIPS log of the secondary server ( from startup, until the moment you want to access a dlg_var ). Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 05/16/2012 08:46 PM, Mariana Arduini wrote: Hi Vlad, Does this also happen if you leave the second server offline, and start it after the active OpenSIPS is shut down (...) ? Yes, that's exactly the test I've run. At the moment that you run dlg_db_sync, do you see the variables in the dialog DB table ? Yes. After you run dlg_db_sync, you say you cannot access the variables from the script, but you see them in dlg_list_ctx ? No, I don't see them in dlg_list_ctx, neither I can access them from the script. Thanks. Mariana. On Wed, May 16, 2012 at 2:31 PM, Vlad Paiu vladp...@opensips.org wrote: Hi Mariana, Does this also happen if you leave the second server offline, and start it after the active OpenSIPS is shut down, instead of leaving the second server up and running 'dlg_db_sync' ? At the moment that you run dlg_db_sync, do you see the variables in the dialog DB table ? After you run dlg_db_sync, you say you cannot access the variables from the script, but you see them in dlg_list_ctx ? Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 05/16/2012 07:57 PM, Mariana Arduini wrote: Hi Razvan, Do I need to open a bug about this issue somewhere? I saw Bogdan's message about OpenSIPS 1.8 Stable being released tomorrow. I think the problem is the dialog variables are not being fetched from DB, either when OpenSIPS is restarded, either when we run the new fifo command dlg_db_sync. Thanks again! Mariana. On Wed, May 16, 2012 at 8:06 AM, Mariana Arduini marianardu...@gmail.com wrote: Hi, Razvan! Thank you for the $DLG_dir pseudovariable, it worked! The variables are properly flushed into the DB after 200 OK, and I can also see them using opensipsctl fifo dlg_list_ctx, under context. Even using the $DLG_dir for the direction of a sequential request, I still need to access either the caller_contact or the callee_contact. Is there any other way to have those apart from the variables? Thanks again! Mariana. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] NAT type
Hola, como puedo detectar cual el tipo de NAT de un cliente? Saludos ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Does OpenSIPS support multithreading?
OpenSIPS 1.8.x Status is Beta released at http://www.opensips.org/Main/Releases but Downloads page http://www.opensips.org/Resources/Downloads says the latest stable release for OpenSIPS is version 1.8.0 1. Does that mean OpenSIPS 1.8.0 is stable and can be used in production environment? 2. Is OpenSIPS 1.8.0 multiprocess and multithreaded? Regards.. VK. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Does OpenSIPS support multithreading?
- OpenSIPS 1.8 was moved from Beta to Stable on May 17, 2012. So yes, it is stable. - OpenSIPS 1.8 is still based on a multi-process architecture. OpenSIPS 2.0 is a architecture re-design version and will be multi-threaded. Regards, Ali On Mon, May 21, 2012 at 10:05 AM, Kumar, Vijendra vku...@softlexicon.comwrote: OpenSIPS 1.8.x Status is Beta released at http://www.opensips.org/Main/Releases but Downloads page http://www.opensips.org/Resources/Downloads says the latest stable release for OpenSIPS is version 1.8.0 1. Does that mean OpenSIPS 1.8.0 is stable and can be used in production environment? 2. Is OpenSIPS 1.8.0 multiprocess and multithreaded? Regards.. VK. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] avpops and oracle
Hi! My opensips server is related to a local mysql database. I want to relay it to an external oracle data base. I do the following, I load the db_oracle module and in opensips.cfg modparam(avpops, db_url, oracle://username:password@hostname/dbname). The strange thing is that when I do restart opensips it restarts and I have the pid number. But actually is not working and my cleints could not register. the login parameter to oracle database are correct, coz when I use sqlplus, I can login. Please, any help. I spent two days in this step and no result :// Thanks a lot ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] kamailio is not relaying BYE message to UAC
Hi, We have the following network architecture : UAC1-kamailioVoipSwitch-PSTN--Phone1 (Sip Client) Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session is terminated cleanly. But if Phone1 hangs up the BYE message which comes to kamailio and goes back to VoipSwitch instead of relayed to UAC1 . So The session becomes a zombie one, And UAC1 unfortunately gets billed for a session which should be terminated. Following is the Call flow as seen from VoipSwitch : | kamilio IP | | | | VoipSwitch IP | |134.856 | INVITE SDP | |(7890) -- (5060) | |134.858 | 407 Proxy Authentication Required | |(7890) -- (5060) | |134.902 | ACK | | |(7890) -- (5060) | |135.408 | INVITE SDP | |(7890) -- (5060) | |135.414 | 100 Trying| | |(7890) -- (5060) | |140.121 | 183 Session Progress SDP | |(7890) -- (5060) | |140.184 | RTP (g729) | | |(61868) -- (5136) | |141.295 | RTP (g729) | | |(61868) -- (5136) | |153.701 | 200 OK SDP | |(7890) -- (5060) | |153.713 | RTP (g729) | | |(61868) -- (5136) | |154.126 | ACK | | |(7890) -- (5060) | |159.988 | BYE | | |(7890) -- (5060) | |160.031 | BYE | | |(7890) -- (5060) | |160.478 | BYE | | |(7890) -- (5060) | |161.412 | BYE | | |(7890) -- (5060) | |181.952 | BYE | | |(7890) -- (5060) | |185.687 | BYE | | |(7890) -- (5060) | |188.018 | 408 Request Timeout | |(7890) -- (5060) | Sip Traces : kamailio--VoipSwitch I'm posting only the offending BYE msg instead of full trace , because of the mail will become difficult to read . If more traces needed, i can post it. The following BYE message is sent by VoipSwitch: BYE sip:ipphone@205.164.40.74 SIP/2.0 Route: sip:108.166.195.189:7890;lr=on;nat=yes CSeq: 2 BYE Via: SIP/2.0/UDP 205.164.40.74:5060 From: sip:008801673345531@205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: 123456 sip:ipphone@205.164.40.74;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 Kamailio instead of relaying the message, sends a BYE message towards VoipSwitch: BYE sip:ipphone@205.164.40.74 SIP/2.0 Max-Forwards: 10 CSeq: 2 BYE Via: SIP/2.0/UDP 108.166.195.189:7890;branch=z9hG4bK4b2b.5d893e95.0 Via: SIP/2.0/UDP 205.164.40.74:5060;rport=5060 From: sip:008801673345531@205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: 123456 sip:ipphone@205.164.40.74;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 When the first BYE message comes from VoipSwitch , kamailio does the following : May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: core [receive.c:289]: receive_msg: cleaning up May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: core [parser/sdp/sdp.c:751]: _sdp = 0x831bf10 May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: core [parser/sdp/sdp.c:753]: sdp = 0x83043dc May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: core [parser/sdp/sdp.c:755]: session = 0x8304504 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: core [parser/msg_parser.c:630]: SIP Request: May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: core [parser/msg_parser.c:632]: method: BYE May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: core [parser/msg_parser.c:634]: uri: sip:ipphone@205.164.40.74 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: core [parser/msg_parser.c:636]: version: SIP/2.0 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: core [parser/msg_parser.c:167]: get_hdr_field: cseq CSeq: 1 BYE May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: core [parser/parse_via.c:1287]: Found param type 232, branch = z9hG4bk19052612230719933454843; state=16 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: core [parser/parse_via.c:2300]: end of header reached, state=5 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: core [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: core [parser/msg_parser.c:517]: parse_headers: this is the first
