[OpenSIPS-Users] Building Telephony Systems with OpenSIPS
Thanks to Flavio E. Goncalves, a new edition of Building Telephony Systems with OpenSIPS is now available. It covers the latest stable release, the OpenSIPS 1.6, updating existing topics (NAT traversal, accounting, etc), but also approaching new 1.6 specific technologies (data caching, dialog usage, etc): The book will teach you how to build scalable and robust telephony systems using SIP: - Build a VoIP Provider based on the SIP Protocol - Cater to scores of subscribers efficiently with a robust telephony system based in pure SIP - Gain a competitive edge using the most scalable VoIP technology - Learn how to avoid pitfalls using precise billing - Packed with rich practical examples and case studies on the latest OpenSIPS version 1.6 More on : http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book Regards, Bogdan -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPs acting as a SIP router only (Sorry about the badly formatted message earlier)
I have just finished installing OpenSIPs about an hour ago. I have gone through the cookbook available but didn't find any elaboration on Source IP rewriting. Before even I start to confuse myself let me explain exactly what I want to achieve with OpenSIPs. I have Asterisk and A2Billing setup properly and they are working fine. OpenSIPs is also installed properly and both are listening on all IPs of the host. I want to use OpenSIPs as a SIP router such that I can send calls and REGISTER to the same host via independent IP addresses. The IPs will be assigned on the basis of the username in the SIP URI. The host has six IPs and all are Public. Can anyone suggest or guide me in the right direction about what module-specific parameters to use or how to use them and a starter script for the routing logic specific to changing source IP. Users will always be behind asterisk and Asterisk will connect to trunks via OpenSIPs. Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs acting as a SIP router only (Sorry about the badly formatted message earlier)
El Jueves, 21 de Enero de 2010, Slot Zero escribió: I have just finished installing OpenSIPs about an hour ago. Perhaps you mean 6 hours ago? XD -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to limit channel on bunch of called DIDs?
I'm facing the same task now - limit the number of concurrent calls per group of accounts rather than a single number. I'm thinking of using the group module to organize numbers into groups with group module, then using get_user_group() to get group id and comparing the profile size with concurrent calls limit set for this group in usr_preferences table. I'd probably hack the get_user_group() function to return the group name instead of id for convenience reason, though. Bogdan-Andrei Iancu wrote: Hi, you do not need any loop - just set as key for profiling the DID number and add to that profile the calls related to that DID. Regards, Bogdan Johnson Pajayat wrote: Hi Bogdan, I was able to implement the channel limiting on one DID by using a variable instead of AVP and replacing all instances of $tU to $rU. Now, I want to limit the channels to a set of DIDs and I'm thinking of implementing a while loop and counter in order to achieve it. Is this an efficient way of doing the limiting on a set of DIDs? One problem I can think with the while loop and counter will be how to deduct those calls that were already hung up by the caller. Again, inputs will be greatly appreciated. Thank you very much. -- Sincerely, Andrew Pogrebennyk ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Iñaki Baz Castillo all wisecrac k but no knowledge
Inaki shutup ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] sched_yield()
Hi Andrei, Hopefully this is it (with FASTLOCK) #0 0xb77d5424 in __kernel_vsyscall () #1 0xb772babb in poll () from /lib/i686/cmov/libc.so.6 #2 0xb77ba83a in ?? () from /lib/i686/cmov/libresolv.so.2 #3 0xb77b8946 in __libc_res_nquery () from /lib/i686/cmov/libresolv.so.2 #4 0xb77b8fdb in ?? () from /lib/i686/cmov/libresolv.so.2 #5 0xb77b92ae in __libc_res_nsearch () from /lib/i686/cmov/libresolv.so.2 #6 0xb77b96d4 in __res_nsearch () from /lib/i686/cmov/libresolv.so.2 #7 0xb77b808a in res_search () from /lib/i686/cmov/libresolv.so.2 #8 0x0808c613 in get_record () #9 0x0808cf05 in ?? () #10 0x0808e385 in sip_resolvehost () #11 0x0807a26c in mk_proxy () #12 0xb7627d39 in t_relay_to () from /usr/lib/opensips/modules/tm.so #13 0xb7634501 in ?? () from /usr/lib/opensips/modules/tm.so #14 0x08055030 in do_action () #15 0x08053ebf in run_action_list () #16 0x08095cf2 in eval_expr () #17 0x080958d9 in eval_expr () #18 0x08095919 in eval_expr () #19 0x080554e2 in do_action () #20 0x08053ebf in run_action_list () #21 0x08056e7a in do_action () #22 0x08053ebf in run_action_list () ---Type return to continue, or q return to quit--- #23 0x080569d8 in do_action () #24 0x08053ebf in run_action_list () #25 0x08056e7a in do_action () #26 0x08053ebf in run_action_list () #27 0x08057d99 in run_top_route () #28 0x0808ad6c in receive_msg () #29 0x080bd2f2 in udp_rcv_loop () #30 0x08069339 in main () (gdb) -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Andrei Dragus Sent: Wednesday, January 20, 2010 2:58 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi, I think that there is a lock that is being held more than it should be and that's what causes starvation. It would help us if you could attach to a process using gdb and give us a full backtrace. Temporary solutions which should work would be to reduce the number of processes to 4-6 or to recompile replacing -DFAST_LOCK with one of the other options (-DUSE_POSIX_SEM or -DUSE_PTHREAD_MUTEX) but we should see where this is from to fix it. Alex Massover wrote: Hi! Yes, from the source on debian, I build deb package. (I did some minor changes to the source, but the problem happens also without my changes) 16 children on 4 cores. What do you suggest to reduce it to 4? It runs on 2.6.32 on VMware ESX. I'm also trying now sleep(0) instead of sched_yield(). -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Andrei Dragus Sent: Wednesday, January 20, 2010 1:05 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi Alex, Are you building OpenSIPS from source? How many processes do you have and on how many cores? Alex Massover wrote: Hello! I'm facing a strange problem, sometimes under a stress OpenSIPS locks - load average jumps, SIP processing delays, opensips msg queue fills with a lot of sip messages, opensips processes start to comsume a lot of CPU. And strace shows: sched_yield() sched_yield() sched_yield() sched_yield() for all processes. If I stop the stress - after a while (not immediately) - it unlocks, also suddenly, I can see in top that all opensips processes stop to consume CPU. What can it be? Some kind of starvation? -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. This mail was sent via Mail-SeCure System. --- -- --- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Andrei Dragus www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users This mail was received via Mail-SeCure System. This mail was sent via Mail-SeCure System. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Andrei Dragus www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users This mail was received via Mail-SeCure System. This mail was sent via Mail-SeCure System. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Iñaki Baz Castillo all wisecrac k but no knowledge
What is this, middle school? In any case, unfortunately for you, you could not be more wrong. Iñaki is one of the most leading experts on SIP protocol mechanics in the open-source community, and his answers are unfailingly precise and detailed. Often, they also contain an element of personality, as is true of any person. This is normal and desirable; it wouldn't be Iñaki (or anyone else in his position) without it. On 01/21/2010 06:50 AM, Slot Zero wrote: Inaki shutup ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Iñaki Baz Castillo all wisecrac k but no knowledge
Well at least he shouldn't have been the one to start it. as far as personality trait well you got two of us; the only difference is that I need loud mouthed wisecracking trigger. So I am still better as far as personality goes. If he is such an expert he could have at least sent a decent reply to my request. I think that is pretty much what I am asking you experts. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] sched_yield()
Hi, Another one.. It hangs for a number of seconds (but it's enough to cause to SIP timeouts - MSG queue jumps to 260K), it's hard to make a bt at the right moment. This one looks better because there's sched_yield() there :) (gdb) bt #0 0xb77d5424 in __kernel_vsyscall () #1 0xb771041c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0x080bf23d in new_avp () #3 0x080bf53f in add_avp () #4 0xb72c1c9c in ?? () from /usr/lib/opensips/modules/dialog.so #5 0x08055030 in do_action () #6 0x08053ebf in run_action_list () #7 0x08056e7a in do_action () #8 0x08053ebf in run_action_list () #9 0x08056e7a in do_action () #10 0x08053ebf in run_action_list () #11 0x08056e7a in do_action () #12 0x08053ebf in run_action_list () #13 0x08057d99 in run_top_route () #14 0x0808ad6c in receive_msg () #15 0x080bd2f2 in udp_rcv_loop () #16 0x08069339 in main () -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Alex Massover Sent: Thursday, January 21, 2010 2:05 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi Andrei, Hopefully this is it (with FASTLOCK) #0 0xb77d5424 in __kernel_vsyscall () #1 0xb772babb in poll () from /lib/i686/cmov/libc.so.6 #2 0xb77ba83a in ?? () from /lib/i686/cmov/libresolv.so.2 #3 0xb77b8946 in __libc_res_nquery () from /lib/i686/cmov/libresolv.so.2 #4 0xb77b8fdb in ?? () from /lib/i686/cmov/libresolv.so.2 #5 0xb77b92ae in __libc_res_nsearch () from /lib/i686/cmov/libresolv.so.2 #6 0xb77b96d4 in __res_nsearch () from /lib/i686/cmov/libresolv.so.2 #7 0xb77b808a in res_search () from /lib/i686/cmov/libresolv.so.2 #8 0x0808c613 in get_record () #9 0x0808cf05 in ?? () #10 0x0808e385 in sip_resolvehost () #11 0x0807a26c in mk_proxy () #12 0xb7627d39 in t_relay_to () from /usr/lib/opensips/modules/tm.so #13 0xb7634501 in ?? () from /usr/lib/opensips/modules/tm.so #14 0x08055030 in do_action () #15 0x08053ebf in run_action_list () #16 0x08095cf2 in eval_expr () #17 0x080958d9 in eval_expr () #18 0x08095919 in eval_expr () #19 0x080554e2 in do_action () #20 0x08053ebf in run_action_list () #21 0x08056e7a in do_action () #22 0x08053ebf in run_action_list () ---Type return to continue, or q return to quit--- #23 0x080569d8 in do_action () #24 0x08053ebf in run_action_list () #25 0x08056e7a in do_action () #26 0x08053ebf in run_action_list () #27 0x08057d99 in run_top_route () #28 0x0808ad6c in receive_msg () #29 0x080bd2f2 in udp_rcv_loop () #30 0x08069339 in main () (gdb) -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Andrei Dragus Sent: Wednesday, January 20, 2010 2:58 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi, I think that there is a lock that is being held more than it should be and that's what causes starvation. It would help us if you could attach to a process using gdb and give us a full backtrace. Temporary solutions which should work would be to reduce the number of processes to 4-6 or to recompile replacing -DFAST_LOCK with one of the other options (-DUSE_POSIX_SEM or -DUSE_PTHREAD_MUTEX) but we should see where this is from to fix it. Alex Massover wrote: Hi! Yes, from the source on debian, I build deb package. (I did some minor changes to the source, but the problem happens also without my changes) 16 children on 4 cores. What do you suggest to reduce it to 4? It runs on 2.6.32 on VMware ESX. I'm also trying now sleep(0) instead of sched_yield(). -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Andrei Dragus Sent: Wednesday, January 20, 2010 1:05 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi Alex, Are you building OpenSIPS from source? How many processes do you have and on how many cores? Alex Massover wrote: Hello! I'm facing a strange problem, sometimes under a stress OpenSIPS locks - load average jumps, SIP processing delays, opensips msg queue fills with a lot of sip messages, opensips processes start to comsume a lot of CPU. And strace shows: sched_yield() sched_yield() sched_yield() sched_yield() for all processes. If I stop the stress - after a while (not immediately) - it unlocks, also suddenly, I can see in top that all opensips processes stop to consume CPU. What can it be? Some kind of starvation? -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. This mail was sent
Re: [OpenSIPS-Users] Iñaki Baz Castillo all wisecrac k but no knowledge
First of all: are you able to use a decent threads capable mail client? all your cool mails appear as a new thread (new conversation) because they don't contain the In-Reply-To header. In order to ask in a maillist you should use a mail client respecting mail threads. El Jueves, 21 de Enero de 2010, Slot Zero escribió: I need loud mouthed wisecracking trigger. So I am still better as far as personality goes. Have you considered paying a consultant for your requeriments? or do you prefer free help/consultancy given exactly as you desire? If he is such an expert he could have at least sent a decent reply to my request. I think that is pretty much what I am asking you experts. After my first reply, which obviously was a joke, I was re-reading your question. But due to your 12-year-old-mind answer I've decided not to waste my time. Good luck. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Iñaki Baz Castillo all wisecrac k but no knowledge
I think Iñaki was trying to tell you politely that your question, was you formulated it, is somewhat naive. This is mainly so because you are looking for a quick answer (as though it could be devised from reading the cookbook) to a potentially rather complicated, larger architectural issues. Certain questions just can't be tackled that way, and certain things can't be learned nor solved in one day, or with a few hours of effort to find a recipe. On 01/21/2010 07:18 AM, Slot Zero wrote: Well at least he shouldn't have been the one to start it. as far as personality trait well you got two of us; the only difference is that I need loud mouthed wisecracking trigger. So I am still better as far as personality goes. If he is such an expert he could have at least sent a decent reply to my request. I think that is pretty much what I am asking you experts. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] sched_yield()
Some more, (gdb) bt #0 0xb78dc424 in __kernel_vsyscall () #1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0xb73d77fd in build_new_dlg () from /usr/lib/opensips/modules/dialog.so #3 0xb73d4b81 in dlg_create_dialog () from /usr/lib/opensips/modules/dialog.so #4 0xb73c9c9e in ?? () from /usr/lib/opensips/modules/dialog.so #5 0x08055030 in do_action () #6 0x08053ebf in run_action_list () #7 0x08056e7a in do_action () #8 0x08053ebf in run_action_list () #9 0x08057d99 in run_top_route () #10 0x0808ad6c in receive_msg () #11 0x080bd2f2 in udp_rcv_loop () #12 0x08069339 in main () (gdb) bt #0 0xb78dc424 in __kernel_vsyscall () #1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0xb77242cd in build_cell () from /usr/lib/opensips/modules/tm.so #3 0xb7739c4a in t_newtran () from /usr/lib/opensips/modules/tm.so #4 0xb772e7b8 in t_relay_to () from /usr/lib/opensips/modules/tm.so #5 0xb773b501 in ?? () from /usr/lib/opensips/modules/tm.so #6 0x08055030 in do_action () #7 0x08053ebf in run_action_list () #8 0x08095cf2 in eval_expr () #9 0x080958d9 in eval_expr () #10 0x08095919 in eval_expr () #11 0x080554e2 in do_action () #12 0x08053ebf in run_action_list () #13 0x080569d8 in do_action () #14 0x08053ebf in run_action_list () #15 0x08056e7a in do_action () #16 0x08053ebf in run_action_list () #17 0x08057d99 in run_top_route () #18 0x0808ad6c in receive_msg () #19 0x080bd2f2 in udp_rcv_loop () #20 0x08069339 in main () -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Alex Massover Sent: Thursday, January 21, 2010 2:24 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi, Another one.. It hangs for a number of seconds (but it's enough to cause to SIP timeouts - MSG queue jumps to 260K), it's hard to make a bt at the right moment. This one looks better because there's sched_yield() there :) (gdb) bt #0 0xb77d5424 in __kernel_vsyscall () #1 0xb771041c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0x080bf23d in new_avp () #3 0x080bf53f in add_avp () #4 0xb72c1c9c in ?? () from /usr/lib/opensips/modules/dialog.so #5 0x08055030 in do_action () #6 0x08053ebf in run_action_list () #7 0x08056e7a in do_action () #8 0x08053ebf in run_action_list () #9 0x08056e7a in do_action () #10 0x08053ebf in run_action_list () #11 0x08056e7a in do_action () #12 0x08053ebf in run_action_list () #13 0x08057d99 in run_top_route () #14 0x0808ad6c in receive_msg () #15 0x080bd2f2 in udp_rcv_loop () #16 0x08069339 in main () -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Alex Massover Sent: Thursday, January 21, 2010 2:05 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi Andrei, Hopefully this is it (with FASTLOCK) #0 0xb77d5424 in __kernel_vsyscall () #1 0xb772babb in poll () from /lib/i686/cmov/libc.so.6 #2 0xb77ba83a in ?? () from /lib/i686/cmov/libresolv.so.2 #3 0xb77b8946 in __libc_res_nquery () from /lib/i686/cmov/libresolv.so.2 #4 0xb77b8fdb in ?? () from /lib/i686/cmov/libresolv.so.2 #5 0xb77b92ae in __libc_res_nsearch () from /lib/i686/cmov/libresolv.so.2 #6 0xb77b96d4 in __res_nsearch () from /lib/i686/cmov/libresolv.so.2 #7 0xb77b808a in res_search () from /lib/i686/cmov/libresolv.so.2 #8 0x0808c613 in get_record () #9 0x0808cf05 in ?? () #10 0x0808e385 in sip_resolvehost () #11 0x0807a26c in mk_proxy () #12 0xb7627d39 in t_relay_to () from /usr/lib/opensips/modules/tm.so #13 0xb7634501 in ?? () from /usr/lib/opensips/modules/tm.so #14 0x08055030 in do_action () #15 0x08053ebf in run_action_list () #16 0x08095cf2 in eval_expr () #17 0x080958d9 in eval_expr () #18 0x08095919 in eval_expr () #19 0x080554e2 in do_action () #20 0x08053ebf in run_action_list () #21 0x08056e7a in do_action () #22 0x08053ebf in run_action_list () ---Type return to continue, or q return to quit--- #23 0x080569d8 in do_action () #24 0x08053ebf in run_action_list () #25 0x08056e7a in do_action () #26 0x08053ebf in run_action_list () #27 0x08057d99 in run_top_route () #28 0x0808ad6c in receive_msg () #29 0x080bd2f2 in udp_rcv_loop () #30 0x08069339 in main () (gdb) -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Andrei Dragus Sent: Wednesday, January 20, 2010 2:58 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi, I think that there is a lock that is being held more than it should be
Re: [OpenSIPS-Users] Iñaki Baz Castillo all wisecrac k but no knowledge
Rather than have this thread continue its downward spiral, I would like to suggest the following book: snip from bogdan's post Thanks to Flavio E. Goncalves, a new edition of Building Telephony Systems with OpenSIPS is now available. It covers the latest stable release, the OpenSIPS 1.6, updating existing topics (NAT traversal, accounting, etc), but also approaching new 1.6 specific technologies (data caching, dialog usage, etc): The book will teach you how to build scalable and robust telephony systems using SIP: - Build a VoIP Provider based on the SIP Protocol - Cater to scores of subscribers efficiently with a robust telephony system based in pure SIP - Gain a competitive edge using the most scalable VoIP technology - Learn how to avoid pitfalls using precise billing - Packed with rich practical examples and case studies on the latest OpenSIPS version 1.6 More on : http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book Regards, Bogdan /snip from bogdan's post Regards, Norm Alex Balashov wrote: I think Iñaki was trying to tell you politely that your question, was you formulated it, is somewhat naive. This is mainly so because you are looking for a quick answer (as though it could be devised from reading the cookbook) to a potentially rather complicated, larger architectural issues. Certain questions just can't be tackled that way, and certain things can't be learned nor solved in one day, or with a few hours of effort to find a recipe. On 01/21/2010 07:18 AM, Slot Zero wrote: Well at least he shouldn't have been the one to start it. as far as personality trait well you got two of us; the only difference is that I need loud mouthed wisecracking trigger. So I am still better as far as personality goes. If he is such an expert he could have at least sent a decent reply to my request. I think that is pretty much what I am asking you experts. ___ 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] A thorough waste of my time fuck you Inaki good Bye
Dear All, I guess none of you have a decent reply as far as my request is concerned. Oh yeah its idiots like Iñaki that prove this place to be a waste of time especially when someone who is trying to get things working. As far as paying consultants dude I am just starting out on my own ok fucking moron. Bogdan I salute you for your efforts but this mother fucker has totally been an asshole and a bad name on your/development group beautiful solution. Thanks to Inaki I will be looking into something else to enable me to do what I want. FUCK YOU INAKI. great waste of my time. Now go fuck yourself. So long. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] sched_yield()
And one more (gdb) bt #0 0xb78dc424 in __kernel_vsyscall () #1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0x080c3705 in _shm_resize () #3 0xb7746069 in relay_reply () from /usr/lib/opensips/modules/tm.so #4 0xb7746d74 in reply_received () from /usr/lib/opensips/modules/tm.so #5 0x08063408 in forward_reply () #6 0x0808ae2b in receive_msg () #7 0x080bd2f2 in udp_rcv_loop () #8 0x08069339 in main () -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Alex Massover Sent: Thursday, January 21, 2010 2:51 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Some more, (gdb) bt #0 0xb78dc424 in __kernel_vsyscall () #1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0xb73d77fd in build_new_dlg () from /usr/lib/opensips/modules/dialog.so #3 0xb73d4b81 in dlg_create_dialog () from /usr/lib/opensips/modules/dialog.so #4 0xb73c9c9e in ?? () from /usr/lib/opensips/modules/dialog.so #5 0x08055030 in do_action () #6 0x08053ebf in run_action_list () #7 0x08056e7a in do_action () #8 0x08053ebf in run_action_list () #9 0x08057d99 in run_top_route () #10 0x0808ad6c in receive_msg () #11 0x080bd2f2 in udp_rcv_loop () #12 0x08069339 in main () (gdb) bt #0 0xb78dc424 in __kernel_vsyscall () #1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0xb77242cd in build_cell () from /usr/lib/opensips/modules/tm.so #3 0xb7739c4a in t_newtran () from /usr/lib/opensips/modules/tm.so #4 0xb772e7b8 in t_relay_to () from /usr/lib/opensips/modules/tm.so #5 0xb773b501 in ?? () from /usr/lib/opensips/modules/tm.so #6 0x08055030 in do_action () #7 0x08053ebf in run_action_list () #8 0x08095cf2 in eval_expr () #9 0x080958d9 in eval_expr () #10 0x08095919 in eval_expr () #11 0x080554e2 in do_action () #12 0x08053ebf in run_action_list () #13 0x080569d8 in do_action () #14 0x08053ebf in run_action_list () #15 0x08056e7a in do_action () #16 0x08053ebf in run_action_list () #17 0x08057d99 in run_top_route () #18 0x0808ad6c in receive_msg () #19 0x080bd2f2 in udp_rcv_loop () #20 0x08069339 in main () -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Alex Massover Sent: Thursday, January 21, 2010 2:24 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi, Another one.. It hangs for a number of seconds (but it's enough to cause to SIP timeouts - MSG queue jumps to 260K), it's hard to make a bt at the right moment. This one looks better because there's sched_yield() there :) (gdb) bt #0 0xb77d5424 in __kernel_vsyscall () #1 0xb771041c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0x080bf23d in new_avp () #3 0x080bf53f in add_avp () #4 0xb72c1c9c in ?? () from /usr/lib/opensips/modules/dialog.so #5 0x08055030 in do_action () #6 0x08053ebf in run_action_list () #7 0x08056e7a in do_action () #8 0x08053ebf in run_action_list () #9 0x08056e7a in do_action () #10 0x08053ebf in run_action_list () #11 0x08056e7a in do_action () #12 0x08053ebf in run_action_list () #13 0x08057d99 in run_top_route () #14 0x0808ad6c in receive_msg () #15 0x080bd2f2 in udp_rcv_loop () #16 0x08069339 in main () -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Alex Massover Sent: Thursday, January 21, 2010 2:05 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi Andrei, Hopefully this is it (with FASTLOCK) #0 0xb77d5424 in __kernel_vsyscall () #1 0xb772babb in poll () from /lib/i686/cmov/libc.so.6 #2 0xb77ba83a in ?? () from /lib/i686/cmov/libresolv.so.2 #3 0xb77b8946 in __libc_res_nquery () from /lib/i686/cmov/libresolv.so.2 #4 0xb77b8fdb in ?? () from /lib/i686/cmov/libresolv.so.2 #5 0xb77b92ae in __libc_res_nsearch () from /lib/i686/cmov/libresolv.so.2 #6 0xb77b96d4 in __res_nsearch () from /lib/i686/cmov/libresolv.so.2 #7 0xb77b808a in res_search () from /lib/i686/cmov/libresolv.so.2 #8 0x0808c613 in get_record () #9 0x0808cf05 in ?? () #10 0x0808e385 in sip_resolvehost () #11 0x0807a26c in mk_proxy () #12 0xb7627d39 in t_relay_to () from /usr/lib/opensips/modules/tm.so #13 0xb7634501 in ?? () from /usr/lib/opensips/modules/tm.so #14 0x08055030 in do_action () #15 0x08053ebf in run_action_list () #16 0x08095cf2 in eval_expr () #17 0x080958d9 in eval_expr () #18 0x08095919 in eval_expr () #19 0x080554e2 in do_action () #20 0x08053ebf in run_action_list () #21 0x08056e7a in do_action ()
Re: [OpenSIPS-Users] sched_yield()
Hi, Since all the backtraces are in allocation routines my guess is that the shared memory lock might be causing a problem. Are you compiling with -DF_MALLOC? What version of OpenSIPS are you using? What is the total shared memory pool you are allocating? What amount of memory are you using? ( Use : opensipsctl fifo get_statistics all ) Alex Massover wrote: Some more, (gdb) bt #0 0xb78dc424 in __kernel_vsyscall () #1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0xb73d77fd in build_new_dlg () from /usr/lib/opensips/modules/dialog.so #3 0xb73d4b81 in dlg_create_dialog () from /usr/lib/opensips/modules/dialog.so #4 0xb73c9c9e in ?? () from /usr/lib/opensips/modules/dialog.so #5 0x08055030 in do_action () #6 0x08053ebf in run_action_list () #7 0x08056e7a in do_action () #8 0x08053ebf in run_action_list () #9 0x08057d99 in run_top_route () #10 0x0808ad6c in receive_msg () #11 0x080bd2f2 in udp_rcv_loop () #12 0x08069339 in main () (gdb) bt #0 0xb78dc424 in __kernel_vsyscall () #1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0xb77242cd in build_cell () from /usr/lib/opensips/modules/tm.so #3 0xb7739c4a in t_newtran () from /usr/lib/opensips/modules/tm.so #4 0xb772e7b8 in t_relay_to () from /usr/lib/opensips/modules/tm.so #5 0xb773b501 in ?? () from /usr/lib/opensips/modules/tm.so #6 0x08055030 in do_action () #7 0x08053ebf in run_action_list () #8 0x08095cf2 in eval_expr () #9 0x080958d9 in eval_expr () #10 0x08095919 in eval_expr () #11 0x080554e2 in do_action () #12 0x08053ebf in run_action_list () #13 0x080569d8 in do_action () #14 0x08053ebf in run_action_list () #15 0x08056e7a in do_action () #16 0x08053ebf in run_action_list () #17 0x08057d99 in run_top_route () #18 0x0808ad6c in receive_msg () #19 0x080bd2f2 in udp_rcv_loop () #20 0x08069339 in main () -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Alex Massover Sent: Thursday, January 21, 2010 2:24 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi, Another one.. It hangs for a number of seconds (but it's enough to cause to SIP timeouts - MSG queue jumps to 260K), it's hard to make a bt at the right moment. This one looks better because there's sched_yield() there :) (gdb) bt #0 0xb77d5424 in __kernel_vsyscall () #1 0xb771041c in sched_yield () from /lib/i686/cmov/libc.so.6 #2 0x080bf23d in new_avp () #3 0x080bf53f in add_avp () #4 0xb72c1c9c in ?? () from /usr/lib/opensips/modules/dialog.so #5 0x08055030 in do_action () #6 0x08053ebf in run_action_list () #7 0x08056e7a in do_action () #8 0x08053ebf in run_action_list () #9 0x08056e7a in do_action () #10 0x08053ebf in run_action_list () #11 0x08056e7a in do_action () #12 0x08053ebf in run_action_list () #13 0x08057d99 in run_top_route () #14 0x0808ad6c in receive_msg () #15 0x080bd2f2 in udp_rcv_loop () #16 0x08069339 in main () -- Best Regards, Alex Massover VoIP RD TL Jajah Inc. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Alex Massover Sent: Thursday, January 21, 2010 2:05 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] sched_yield() Hi Andrei, Hopefully this is it (with FASTLOCK) #0 0xb77d5424 in __kernel_vsyscall () #1 0xb772babb in poll () from /lib/i686/cmov/libc.so.6 #2 0xb77ba83a in ?? () from /lib/i686/cmov/libresolv.so.2 #3 0xb77b8946 in __libc_res_nquery () from /lib/i686/cmov/libresolv.so.2 #4 0xb77b8fdb in ?? () from /lib/i686/cmov/libresolv.so.2 #5 0xb77b92ae in __libc_res_nsearch () from /lib/i686/cmov/libresolv.so.2 #6 0xb77b96d4 in __res_nsearch () from /lib/i686/cmov/libresolv.so.2 #7 0xb77b808a in res_search () from /lib/i686/cmov/libresolv.so.2 #8 0x0808c613 in get_record () #9 0x0808cf05 in ?? () #10 0x0808e385 in sip_resolvehost () #11 0x0807a26c in mk_proxy () #12 0xb7627d39 in t_relay_to () from /usr/lib/opensips/modules/tm.so #13 0xb7634501 in ?? () from /usr/lib/opensips/modules/tm.so #14 0x08055030 in do_action () #15 0x08053ebf in run_action_list () #16 0x08095cf2 in eval_expr () #17 0x080958d9 in eval_expr () #18 0x08095919 in eval_expr () #19 0x080554e2 in do_action () #20 0x08053ebf in run_action_list () #21 0x08056e7a in do_action () #22 0x08053ebf in run_action_list () ---Type return to continue, or q return to quit--- #23 0x080569d8 in do_action () #24 0x08053ebf in run_action_list () #25 0x08056e7a in do_action () #26 0x08053ebf in run_action_list () #27 0x08057d99 in run_top_route () #28 0x0808ad6c in receive_msg () #29 0x080bd2f2 in udp_rcv_loop () #30 0x08069339 in main () (gdb) -- Best Regards, Alex Massover VoIP RD TL Jajah
Re: [OpenSIPS-Users] A thorough waste of my time fuck you Inaki good Bye
Hey Kiddo, Someday, when you grow up, you may realise that someone was just trying to tell you - in a rather sagacious, terse manner - that the question you were asking was not really of a manageable or addressable scope, nor especially coherent. You're actually much better off that way, rather than by having your question interrogated on a much more granular level by people who have vastly more experience and accumulated knowledge. Check this document out: http://catb.org/~esr/faqs/smart-questions.html In your case, I'm specifically thinking: http://catb.org/~esr/faqs/smart-questions.html#volume http://catb.org/~esr/faqs/smart-questions.html#goal http://catb.org/~esr/faqs/smart-questions.html#explicit -- Alex On 01/21/2010 07:58 AM, Slot Zero wrote: Dear All, I guess none of you have a decent reply as far as my request is concerned. Oh yeah its idiots like Iñaki that prove this place to be a waste of time especially when someone who is trying to get things working. As far as paying consultants dude I am just starting out on my own ok fucking moron. Bogdan I salute you for your efforts but this mother fucker has totally been an asshole and a bad name on your/development group beautiful solution. Thanks to Inaki I will be looking into something else to enable me to do what I want. FUCK YOU INAKI. great waste of my time. Now go fuck yourself. So long. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Private IP in registered AOR causing failure
Hello Bogdan, An mer., janv 20, 2010, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: Here's a record I see when I run 'opensipsctl ul show': AOR:: mylogin-osips Contact:: sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm Q=1 Expires:: 560 Callid:: 2b21cdfae784-av13rj1txbsq Cseq:: 2 User-agent:: Bigphone123 Received:: sip:85.182.68.45:2240;transport=TLS State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: tls:80.200.123.45:5061 Methods:: 7999 OpenSIPS is trying to reach the private IP number above from time to time, and I see this in the logs: Jan 19 17:57:20 name.host.tld error opensips[23432]: ERROR:tm:t_uac: attempt to send to 'sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm' failed Thanks for any advice on correcting the failed private IP attempts. the problem is not the private IP in the contact, as opensips properly saved the source IP (of the REGISTER) too - see the Received field. So the Received field will be used over the Contact for sending the requests to UAC. Yes, that's what was confusing me, that the Received header was correct and the TCP connection still failed. Now, what probably goes wrong in your case is that when using TLS/TCP (connection oriented protos), after the REGISTER, the connection is dropped and opensips cannot open later a TCP connection behind a NAT :(By default opensips closes the inactive TCP connections. To make opensips to keep the connection (even with no traffic going on), see the tcp_persistent_flag: http://www.opensips.org/html/docs/modules/devel/registrar.html#id228181 Good guess, but sadly your're wrong. I'm well informed of manipulating the TCP connection using the tcp_persistent_flag. It just wasn't where you expected it (and didn't appear in my email.) The tcp_persistent_flag is being set on all registrations just before save(). When I use tcpdump to see which address OpenSIPS is trying to connect to before reporting the failure above, I see that it really is trying to connect to 192.168.0.31 (over the Internet.) If I write fix_contact() just before save() then this behaviour and the failure goes away. It would seem that a lookup(location) does not always use the AOR's Received header, or what do you think could be the problem? It doesn't seem right changing the AOR's contact string to hack this problem away. Greetings, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Nat_traversal version of fix_nated_sdp()
Hello Bogdan, An mer., janv 20, 2010, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: Because nat_traversal is newer and seems to be where most of the NAT logic will be developed in the future, it seems the natural choice. If I'm right about that, That's not really true. nathelper is used together with rtpproxy (as media rely), while nat_traversal in used together with mediaproxy. Both are to be developed as alternative solutions to the same problem (at least for the moment) Thanks for clearing that up. It seems strange that so many features of both modules indent to do the same thing. How do I stop using the fix_nated_sdp() function from nathelper? Is there such a function in the nat_traversal module, or is there another way to achieve the same thing without using nathelper? nat_traversal module provides support only for signalling part (afaik) and not for media/SDP part. If you need the functionality of fix_nated_sdp(), you need nathelper module. Okay, I'll go back to using the nathelper module. The mediaproxy and rtpproxy components don't interest me at all, though. Greetings, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] sched_yield()
Hi!
Yes, with -DF_MALLOC.
1.6.1 from sources, I build deb package.
I use 128M of shared and 10*1024*1024 private memory (can increase - no
problem).
H, opensipsctl fifo get_statistics all crashes/stops the opensips.
'fifo uptime' or 'fifo debug' are OK.
strace while 'fifo get_statistics all':
Process 9509 attached - interrupt to quit
pause() = ? ERESTARTNOHAND (To be restarted)
--- SIGUSR2 (User defined signal 2) @ 0 (0) ---
sigreturn() = ? (mask now [])
pause() = ? ERESTARTNOHAND (To be restarted)
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
waitpid(-1, [{WIFSIGNALED(s) WTERMSIG(s) == SIGUSR2}], WNOHANG) = 9520
waitpid(-1, 0xbf84b4c8, WNOHANG)= 0
kill(0, SIGTERM)= 0
--- SIGTERM (Terminated) @ 0 (0) ---
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [TERM])
sigreturn() = ? (mask now [])
rt_sigaction(SIGALRM, {0x8065920, [ALRM], SA_RESTART}, {SIG_DFL}, 8) = 0
alarm(60) = 0
wait4(-1, NULL, 0, NULL)= 9514
wait4(-1, NULL, 0, NULL)= 9519
wait4(-1, NULL, 0, NULL)= 9521
wait4(-1, NULL, 0, NULL)= 9522
wait4(-1, NULL, 0, NULL)= 9512
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
wait4(-1, NULL, 0, NULL)= 9510
wait4(-1, NULL, 0, NULL)= 9516
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
wait4(-1, NULL, 0, NULL)= 9515
wait4(-1, NULL, 0, NULL)= 9517
wait4(-1, NULL, 0, NULL)= 9524
wait4(-1, NULL, 0, NULL)= 9525
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
wait4(-1, NULL, 0, NULL)= 9511
wait4(-1, NULL, 0, NULL)= 9513
wait4(-1, NULL, 0, NULL)= 9518
wait4(-1, NULL, 0, NULL)= 9523
wait4(-1, NULL, 0, NULL)= -1 ECHILD (No child processes)
rt_sigaction(SIGALRM, {0x8066080, [ALRM], SA_RESTART}, {0x8065920, [ALRM],
SA_RESTART}, 8) = 0
stat64(/tmp/opensips_fifo, {st_mode=S_IFIFO|0660, st_size=0, ...}) = 0
unlink(/tmp/opensips_fifo)= 0
munmap(0xaed25000, 134217728) = 0
unlink(/var/run/opensips/opensips.pid) = 0
alarm(0)= 60
rt_sigaction(SIGALRM, {SIG_IGN}, {0x8066080, [ALRM], SA_RESTART}, 8) = 0
exit_group(0) = ?
Process 9509 detached
--
Best Regards,
Alex Massover
VoIP RD TL
Jajah Inc.
-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Andrei Dragus
Sent: Thursday, January 21, 2010 3:09 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] sched_yield()
Hi,
Since all the backtraces are in allocation routines my guess is that
the
shared memory lock might be causing a problem.
Are you compiling with -DF_MALLOC?
What version of OpenSIPS are you using?
What is the total shared memory pool you are allocating?
What amount of memory are you using? ( Use : opensipsctl fifo
get_statistics all )
Alex Massover wrote:
Some more,
(gdb) bt
#0 0xb78dc424 in __kernel_vsyscall ()
#1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6
#2 0xb73d77fd in build_new_dlg () from
/usr/lib/opensips/modules/dialog.so
#3 0xb73d4b81 in dlg_create_dialog () from
/usr/lib/opensips/modules/dialog.so
#4 0xb73c9c9e in ?? () from /usr/lib/opensips/modules/dialog.so
#5 0x08055030 in do_action ()
#6 0x08053ebf in run_action_list ()
#7 0x08056e7a in do_action ()
#8 0x08053ebf in run_action_list ()
#9 0x08057d99 in run_top_route ()
#10 0x0808ad6c in receive_msg ()
#11 0x080bd2f2 in udp_rcv_loop ()
#12 0x08069339 in main ()
(gdb) bt
#0 0xb78dc424 in __kernel_vsyscall ()
#1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6
#2 0xb77242cd in build_cell () from /usr/lib/opensips/modules/tm.so
#3 0xb7739c4a in t_newtran () from /usr/lib/opensips/modules/tm.so
#4 0xb772e7b8 in t_relay_to () from /usr/lib/opensips/modules/tm.so
#5 0xb773b501 in ?? () from /usr/lib/opensips/modules/tm.so
#6 0x08055030 in do_action ()
#7 0x08053ebf in
[OpenSIPS-Users] OpenSIPS Crashed!!
Hi, Right now I am using OpenSIPS 1.5.3 no-tls version in production. Suddenly OpenSIPS got crashed. I did check coredump but can not understand it. Please help me out to interpret it. Here is the pastebin link which contains output of bt. http://pastebin.com/m49520853 Thanks in advance!!! -- VoipExpert ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] sched_yield()
My guess is that there is not enough shared memory. When an allocation
failes OpenSIPS tries to defragment memory to make room which takes a
lot of time and must be done under lock.
Please try to increase the shared memory size and tell me if it persists.
Alex Massover wrote:
Hi!
Yes, with -DF_MALLOC.
1.6.1 from sources, I build deb package.
I use 128M of shared and 10*1024*1024 private memory (can increase - no
problem).
H, opensipsctl fifo get_statistics all crashes/stops the opensips.
'fifo uptime' or 'fifo debug' are OK.
strace while 'fifo get_statistics all':
Process 9509 attached - interrupt to quit
pause() = ? ERESTARTNOHAND (To be restarted)
--- SIGUSR2 (User defined signal 2) @ 0 (0) ---
sigreturn() = ? (mask now [])
pause() = ? ERESTARTNOHAND (To be restarted)
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
waitpid(-1, [{WIFSIGNALED(s) WTERMSIG(s) == SIGUSR2}], WNOHANG) = 9520
waitpid(-1, 0xbf84b4c8, WNOHANG)= 0
kill(0, SIGTERM)= 0
--- SIGTERM (Terminated) @ 0 (0) ---
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [TERM])
sigreturn() = ? (mask now [])
rt_sigaction(SIGALRM, {0x8065920, [ALRM], SA_RESTART}, {SIG_DFL}, 8) = 0
alarm(60) = 0
wait4(-1, NULL, 0, NULL)= 9514
wait4(-1, NULL, 0, NULL)= 9519
wait4(-1, NULL, 0, NULL)= 9521
wait4(-1, NULL, 0, NULL)= 9522
wait4(-1, NULL, 0, NULL)= 9512
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
wait4(-1, NULL, 0, NULL)= 9510
wait4(-1, NULL, 0, NULL)= 9516
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
wait4(-1, NULL, 0, NULL)= 9515
wait4(-1, NULL, 0, NULL)= 9517
wait4(-1, NULL, 0, NULL)= 9524
wait4(-1, NULL, 0, NULL)= 9525
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
wait4(-1, NULL, 0, NULL)= 9511
wait4(-1, NULL, 0, NULL)= 9513
wait4(-1, NULL, 0, NULL)= 9518
wait4(-1, NULL, 0, NULL)= 9523
wait4(-1, NULL, 0, NULL)= -1 ECHILD (No child processes)
rt_sigaction(SIGALRM, {0x8066080, [ALRM], SA_RESTART}, {0x8065920, [ALRM],
SA_RESTART}, 8) = 0
stat64(/tmp/opensips_fifo, {st_mode=S_IFIFO|0660, st_size=0, ...}) = 0
unlink(/tmp/opensips_fifo)= 0
munmap(0xaed25000, 134217728) = 0
unlink(/var/run/opensips/opensips.pid) = 0
alarm(0)= 60
rt_sigaction(SIGALRM, {SIG_IGN}, {0x8066080, [ALRM], SA_RESTART}, 8) = 0
exit_group(0) = ?
Process 9509 detached
--
Best Regards,
Alex Massover
VoIP RD TL
Jajah Inc.
-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Andrei Dragus
Sent: Thursday, January 21, 2010 3:09 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] sched_yield()
Hi,
Since all the backtraces are in allocation routines my guess is that
the
shared memory lock might be causing a problem.
Are you compiling with -DF_MALLOC?
What version of OpenSIPS are you using?
What is the total shared memory pool you are allocating?
What amount of memory are you using? ( Use : opensipsctl fifo
get_statistics all )
Alex Massover wrote:
Some more,
(gdb) bt
#0 0xb78dc424 in __kernel_vsyscall ()
#1 0xb781741c in sched_yield () from /lib/i686/cmov/libc.so.6
#2 0xb73d77fd in build_new_dlg () from
/usr/lib/opensips/modules/dialog.so
#3 0xb73d4b81 in dlg_create_dialog () from
/usr/lib/opensips/modules/dialog.so
#4 0xb73c9c9e in ?? () from /usr/lib/opensips/modules/dialog.so
#5 0x08055030 in do_action ()
#6 0x08053ebf in run_action_list ()
#7 0x08056e7a in do_action ()
#8 0x08053ebf in run_action_list ()
#9 0x08057d99 in run_top_route ()
#10 0x0808ad6c in receive_msg ()
#11 0x080bd2f2 in udp_rcv_loop ()
#12 0x08069339 in main ()
(gdb) bt
#0 0xb78dc424 in __kernel_vsyscall ()
#1 0xb781741c in
Re: [OpenSIPS-Users] INVITE not forwarded, call fails
On 09/12/09 18:09, Bogdan-Andrei Iancu wrote: Hi Lorenzo, check with opensipsctl ul show how your phone is registered - what is important are the contact and received fields - which is present and which contains private IPs. Hi Bogdan! first of all, thanks for the very late reply, been very busy lately! i've checked with opensipsctl ul show, and what shows up is that my app is recorded with the wrong port! i.e. not the one in the ip header, but the one in the sip header.! it's as if rport was ignored! AOR:: asymmetric Contact:: sip:asymmet...@151.16.109.231:62560;transport=udp Q= Expires:: 3565 Callid:: 445710225...@151.16.109.231 Cseq:: 1 User-agent:: mjsip stack 1.6 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:88.149.188.29:5060 Methods:: 4294967295 62560 is wrong, while the ip is correct (which means that stun works correctly) i probably misconfigured opensips.cfg, could you give me some advice? here's mine, in all its glory ;) http://paste2.org/p/623910 btw, i'm running a very basic setup with no db, is that a problem? thanks a lot, lorenzo Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Need Help on integrating opensips with mysql on remote machine
yes, with same user-name and passwd i am able to login from opensips box to remote mysql server. -- View this message in context: http://n2.nabble.com/Need-Help-on-integrating-opensips-with-mysql-on-remote-machine-tp4287536p4433694.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] problem with uac registration
On 16/11/09 15:49, Bogdan-Andrei Iancu wrote: Hi Lorenzo, Check the followings: (1) your REGISTERS hits opensips (run an ngrep on opensips's machine) (2) check where the reply from opensips is sent to (maybe it is sent to wrong destination) hi bogdan! again, sorry for the late reply! (1) yes, they do hit opensips (2) there's no reply for what i can see.. this is the parameters i'm feeding to ngrep: sudo ngrep -tW byline port 5060 should be ok, right? if you need some excerpts of the debug log, just let me know (along with the debug level needed) Regards, Bogdan thanks again, lorenzo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] media proxy - B2BUA(signaling only)
I'm having a problem that I think is the same as this discussion. When a call
from a natted user comes in to my opensips proxy, my config does auth and then
immediately fires up the nat ping and media proxy ( engage-mediaproxy() ) to
provide far end nat traversal. Now I'm trying to add top hiding from b2bua and
when the INVITE comes from b2bua back to opensips, the sdp has values from the
original invite rather than the mediaproxy IP and port.
I suspect that engaging the media proxy on the way out of the proxy rather than
on the inbound side will resolve this, although I haven't tried that yet.
However, this doesn't seem like expected behavior from the b2bua so I thought I
would chime in here.
Richard
sidenoteIñaki, making new friends i see. : /sidenote
On Jan 20, 2010, at 3:37 PM, Julien Chavanton wrote:
Hi Bogdan,
I have found this logic that is working (tested), I think it will not create
problem as it is not possible for a new INVITE to already have the
media-proxy IP if is not already passing trought the proxy once.
#--- Force media proxy
if(is_present_hf(Media-Proxy)){
xlog(L_NOTICE, ***MEDIA PROXY FOUND [on another call leg]\n***);
}
else{
if(search_body(c=IN IP4 1.1.1.1) method == INVITE){
#--- Tag the call, not to engage media proxy
append_hf(Media-Proxy: engaged\r\n);
}
else{
$avp(s:media_relay) = 1.1.1.1;
use_media_proxy();
}
}
From: users-boun...@lists.opensips.org on behalf of Bogdan-Andrei Iancu
Sent: Wed 20/01/2010 5:58 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] media proxy - B2BUA(signaling only)
Hi Julien,
Just a wild guess - maybe the mediaproxy does not work as you may invoke
it (request and reply) for different calls (for request on UA_A call and
for reply on UA_B calls) - afaik, both mediaproxy and rtpproxy use the
callid in order to bridge the request and reply SDP portsBut because
of B2BUA, you actually have 2 calls.
So, be sure you do both media setup steps on the same call (only on one
side of the B2BUA).
Regards,
Bogdan
Julien Chavanton wrote:
Humm, this does not work, as when the 200 OK comes back from UA_B
there is a need to set media proxy and a second session is set on the
media proxy.
Any suggestion on how to handle this senario ?
*From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton
*Sent:* Tue 19/01/2010 8:55 PM
*To:* OpenSIPS users mailling list; Users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] media proxy - B2BUA(signaling only)
I found what was going on more in details, when the second call comes
in the RTP port is set as the correct one from the first call,
however, I issue another use_media_proxy() and a new port is set, I
guess I could ignore the use media proxy this time by validating if
the IP of the media proxy is already in SDP m=audio x.x.x.x, I will
try this
Opensip_A --invite-- B2B_UA
m=audio 52542 RTP/AVP 18 8 0
B2B_UA --invite-- Opensip_A
m=audio 52542 RTP/AVP 18 8 0
(use media proxy)
Opensip_A --invite-- UA_B
m=audio 50506 RTP/AVP 18 8 0
*From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton
*Sent:* Tue 19/01/2010 6:43 PM
*To:* Users@lists.opensips.org
*Subject:* [OpenSIPS-Users] media proxy - B2BUA(signaling only)
Hi,
I have this senario :
UA_A -- Opensip_A -- SIP_B2BUA(signaling only) -- Opensip_A -- UA_B
this result is 2 calls(different callid) on Opensip_A however UA_A and
UA_B are connecting the the MediaProxy on 2 differents calls and the
audio is not bridged between them.
It there a way the make the media proxy aware about the fact that the
2 calls should be bridged ?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
winmail.dat___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Private IP in registered AOR causing failure
Hello again, An mer., janv 20, 2010, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: Here's a record I see when I run 'opensipsctl ul show': AOR:: mylogin-osips Contact:: sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm Q=1 Expires:: 560 Callid:: 2b21cdfae784-av13rj1txbsq Cseq:: 2 User-agent:: Bigphone123 Received:: sip:85.182.68.45:2240;transport=TLS State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: tls:80.200.123.45:5061 Methods:: 7999 OpenSIPS is trying to reach the private IP number above from time to time, and I see this in the logs: Jan 19 17:57:20 name.host.tld error opensips[23432]: ERROR:tm:t_uac: attempt to send to 'sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm' failed the problem is not the private IP in the contact, as opensips properly saved the source IP (of the REGISTER) too - see the Received field. So the Received field will be used over the Contact for sending the requests to UAC. Now, what probably goes wrong in your case is that when using TLS/TCP (connection oriented protos), after the REGISTER, the connection is dropped and opensips cannot open later a TCP connection behind a NAT :(By default opensips closes the inactive TCP connections. After running a socket listener on 192.168.0.31 on the OpenSIPS host: $ socat TCP4-LISTEN:2310,bind=192.168.0.31,reuseaddr - SUBSCRIBE sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm SIP/2.0 Via: SIP/2.0/TCP 86.90.39.44;branch=G4z9hb82dK8.f144.0 To: sip:mylogin-os...@name.host.tld;tag=ty6sjh9iz9 From: sip:mylogin-os...@name.host.tld;tag=6c9d4319c74d756e6b696-baa1 CSeq: 11 SUBSCRIBE Call-ID: b1c04118-8...@86.90.39.44 Content-Length: 0 User-Agent: OpenSIPS (1.6.1-tls) Max-Forwards: 70 Event: dialog;sla Contact: sip:prese...@name.host.tld Expires: 610 I'm trying to implement presence by using the presence, presence_xml, pua, and pua_bla modules. So it seems that one of these modules (see event dialog;sla) is getting the contact from the locations table (in AAA on our server) and ignoring the Received header. OpenSIPS replies to messages from UACs such as INVITE and CANCEL properly, and opens connections to the IP in Received. This problem is limited to the SUBSCRIBES sent from one of the presence modules. ...but I'm still not sure how to fix the problem. Any ideas? Greetings, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Distinction between two forked INVITE received from upstream
Hello, The next figure describes my problem. openSIPS +--+ -- INVITE(1) --| |--- INVITE(1) -- | | -- INVITE(2) --| |--- INVITE(2) -- -- 180(2) --| |-- 180(2) -- -- 200(2) --| |-- 200(2) -- | | -- CANCEL(1) --| | -- 200(CANCEL) -| | -- 487(1) --| | -- ACK(1) -| | | | -- ACK(2) -| |--- ACK(2) - | | +--+ | | +--+ | | | | +--+ RTP proxy I use RTP proxy to relay media streams. INVITE (1) and (2) have been forked from an upstream proxy, thus they have the same Call-ID and From tag, and a different branch ID. INVITE (2) is accepted, but INVITE (1) has no response. The upstream proxy cancels INVITE (1). In my script, on CANCEL(1) request I close the RTP session, but I should not dot that since INVITE(2) has been accepted. My questions are: How can I know when I received CANCEL(1) that INVITE(2) has been accepted? Do I need to use flags? Which ones ? Thanks for any help, Yannick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] A thorough waste of my time fuck you Inaki good Bye
I'd just like to say that I for one appreciate both Inaki's participation in the group and his wit. If you can't take it, then perhaps you'd do better off learning on your own. We are all happy to help (after all, we participate because we enjoy it... well I do anyway), but we arn't going to do the work for you. On Thu, Jan 21, 2010 at 6:58 AM, Slot Zero slotze...@yahoo.com wrote: Dear All, I guess none of you have a decent reply as far as my request is concerned. Oh yeah its idiots like Iñaki that prove this place to be a waste of time especially when someone who is trying to get things working. As far as paying consultants dude I am just starting out on my own ok fucking moron. Bogdan I salute you for your efforts but this mother fucker has totally been an asshole and a bad name on your/development group beautiful solution. Thanks to Inaki I will be looking into something else to enable me to do what I want. FUCK YOU INAKI. great waste of my time. Now go fuck yourself. So long. ___ 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] Distinction between two forked INVITE received from upstream
Hi Yannick, indeed the issue is tricky as the media relay uses as a key (for the streams) the SIP callid+from_tag+to_tag(optional) . so, it an early stage, when to_tag is not yet know, the media relay cannot make difference between the two branches of the same call. My advice - do not do stop for media relay on CANCEL or call setup fail as you never know how many parallel branches may belet is timeout is indeed the call was completely ended. Regards, Bogdan Yannick LE COENT wrote: Hello, The next figure describes my problem. openSIPS +--+ -- INVITE(1) --| |--- INVITE(1) -- | | -- INVITE(2) --| |--- INVITE(2) -- -- 180(2) --| |-- 180(2) -- -- 200(2) --| |-- 200(2) -- | | -- CANCEL(1) --| | -- 200(CANCEL) -| | -- 487(1) --| | -- ACK(1) -| | | | -- ACK(2) -| |--- ACK(2) - | | +--+ | | +--+ | | | | +--+ RTP proxy I use RTP proxy to relay media streams. INVITE (1) and (2) have been forked from an upstream proxy, thus they have the same Call-ID and From tag, and a different branch ID. INVITE (2) is accepted, but INVITE (1) has no response. The upstream proxy cancels INVITE (1). In my script, on CANCEL(1) request I close the RTP session, but I should not dot that since INVITE(2) has been accepted. My questions are: How can I know when I received CANCEL(1) that INVITE(2) has been accepted? Do I need to use flags? Which ones ? Thanks for any help, Yannick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Distinction between two forked INVITE received from upstream
This is an interesting question. The requests should have different tags .I'm not sure how OpenSIPs handles the individual requests.. I'd tend to expect a 491 Request Pending or a 500 Requests Merged. But maybe that's B2BUA behavior. Since opensips isn't replying to the second INVITE, it's almost like it's absorbing it as a retransmission. I'm not sure what your expected behavior is, but it seems that you're going to constantly have race conditions on this and that the outcome will be random (ie: does INVITE(1) hit first, or INVITE(2)) What's the larger picture here? Why do you have two INVITEs from the same call hitting your proxy? -Brett On Thu, Jan 21, 2010 at 8:50 AM, Yannick LE COENT yannick.leco...@nexcom.fr wrote: Hello, The next figure describes my problem. openSIPS +--+ -- INVITE(1) --| |--- INVITE(1) -- | | -- INVITE(2) --| |--- INVITE(2) -- -- 180(2) --| |-- 180(2) -- -- 200(2) --| |-- 200(2) -- | | -- CANCEL(1) --| | -- 200(CANCEL) -| | -- 487(1) --| | -- ACK(1) -| | | | -- ACK(2) -| |--- ACK(2) - | | +--+ | | +--+ | | | | +--+ RTP proxy I use RTP proxy to relay media streams. INVITE (1) and (2) have been forked from an upstream proxy, thus they have the same Call-ID and From tag, and a different branch ID. INVITE (2) is accepted, but INVITE (1) has no response. The upstream proxy cancels INVITE (1). In my script, on CANCEL(1) request I close the RTP session, but I should not dot that since INVITE(2) has been accepted. My questions are: How can I know when I received CANCEL(1) that INVITE(2) has been accepted? Do I need to use flags? Which ones ? Thanks for any help, Yannick ___ 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] Need Help on integrating opensips with mysql on remote machine
Just for grins, can you try adding an entry in your /etc/hosts file for your mysql server address? The error specifically says it can't reverse resolve it: Jan 11 07:01:15 localhost opensips: WARNING:core:fix_socket_list: could not rev. resolve 200.200.100.11 BTW, is that really the IP you are using? and is it real -Brett On Thu, Jan 21, 2010 at 7:50 AM, Alok Kushwaha alok.c...@gmail.com wrote: yes, with same user-name and passwd i am able to login from opensips box to remote mysql server. -- View this message in context: http://n2.nabble.com/Need-Help-on-integrating-opensips-with-mysql-on-remote-machine-tp4287536p4433694.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ 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] media proxy - B2BUA(signaling only)
Hi Richard, the solution I am using is working, but the header was useless, the
only problem is that a second media_proxy session is still opened, but this is
not an issue
#--- Force media proxy
if(search_body(c=IN IP4 1.1.1.1) method == INVITE){
xlog(L_NOTICE, ***MEDIA PROXY FOUND [on another call
leg]\n***);
}
else{
$avp(s:media_relay) = 1.1.1.1;
use_media_proxy();
}
From: users-boun...@lists.opensips.org on behalf of Richard Revels
Sent: Thu 21/01/2010 2:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] media proxy - B2BUA(signaling only)
I'm having a problem that I think is the same as this discussion. When a call
from a natted user comes in to my opensips proxy, my config does auth and then
immediately fires up the nat ping and media proxy ( engage-mediaproxy() ) to
provide far end nat traversal. Now I'm trying to add top hiding from b2bua and
when the INVITE comes from b2bua back to opensips, the sdp has values from the
original invite rather than the mediaproxy IP and port.
I suspect that engaging the media proxy on the way out of the proxy rather than
on the inbound side will resolve this, although I haven't tried that yet.
However, this doesn't seem like expected behavior from the b2bua so I thought I
would chime in here.
Richard
sidenoteIñaki, making new friends i see. : /sidenote
On Jan 20, 2010, at 3:37 PM, Julien Chavanton wrote:
Hi Bogdan,
I have found this logic that is working (tested), I think it will not create
problem as it is not possible for a new INVITE to already have the
media-proxy IP if is not already passing trought the proxy once.
#--- Force media proxy
if(is_present_hf(Media-Proxy)){
xlog(L_NOTICE, ***MEDIA PROXY FOUND [on another call leg]\n***);
}
else{
if(search_body(c=IN IP4 1.1.1.1) method == INVITE){
#--- Tag the call, not to engage media proxy
append_hf(Media-Proxy: engaged\r\n);
}
else{
$avp(s:media_relay) = 1.1.1.1;
use_media_proxy();
}
}
From: users-boun...@lists.opensips.org on behalf of Bogdan-Andrei Iancu
Sent: Wed 20/01/2010 5:58 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] media proxy - B2BUA(signaling only)
Hi Julien,
Just a wild guess - maybe the mediaproxy does not work as you may invoke
it (request and reply) for different calls (for request on UA_A call and
for reply on UA_B calls) - afaik, both mediaproxy and rtpproxy use the
callid in order to bridge the request and reply SDP portsBut because
of B2BUA, you actually have 2 calls.
So, be sure you do both media setup steps on the same call (only on one
side of the B2BUA).
Regards,
Bogdan
Julien Chavanton wrote:
Humm, this does not work, as when the 200 OK comes back from UA_B
there is a need to set media proxy and a second session is set on the
media proxy.
Any suggestion on how to handle this senario ?
*From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton
*Sent:* Tue 19/01/2010 8:55 PM
*To:* OpenSIPS users mailling list; Users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] media proxy - B2BUA(signaling only)
I found what was going on more in details, when the second call comes
in the RTP port is set as the correct one from the first call,
however, I issue another use_media_proxy() and a new port is set, I
guess I could ignore the use media proxy this time by validating if
the IP of the media proxy is already in SDP m=audio x.x.x.x, I will
try this
Opensip_A --invite-- B2B_UA
m=audio 52542 RTP/AVP 18 8 0
B2B_UA --invite-- Opensip_A
m=audio 52542 RTP/AVP 18 8 0
(use media proxy)
Opensip_A --invite-- UA_B
m=audio 50506 RTP/AVP 18 8 0
*From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton
*Sent:* Tue 19/01/2010 6:43 PM
*To:* Users@lists.opensips.org
*Subject:* [OpenSIPS-Users] media proxy - B2BUA(signaling only)
Hi,
I have this senario :
UA_A -- Opensip_A -- SIP_B2BUA(signaling only) -- Opensip_A -- UA_B
this result is 2 calls(different callid) on Opensip_A however UA_A and
UA_B are connecting the the MediaProxy on 2 differents calls and the
audio is not bridged between them.
It there a way the make the media proxy aware about the fact that the
2 calls should be bridged ?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users
[OpenSIPS-Users] ping gateways in lcr or drouting?
Hi, Is there any feature in drouting or lcr module to stop selecting a failing gw for a particular amount of time? It would suffice to have a ping mechanism or alternatively, mark the gateway as defunct in the failure_route. I was thinking pinging mechanism is present at least in lcr, but right now I see it only in dispatcher module. -- Sincerely, Andrew Pogrebennyk ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ping gateways in lcr or drouting?
Hi Andrew, None of them do support pinging to GW, but I guess it will be a nice feature for DR.. Regards, Bogdan Andrew Pogrebennyk wrote: Hi, Is there any feature in drouting or lcr module to stop selecting a failing gw for a particular amount of time? It would suffice to have a ping mechanism or alternatively, mark the gateway as defunct in the failure_route. I was thinking pinging mechanism is present at least in lcr, but right now I see it only in dispatcher module. -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Music On Hold Opensips
Hello, I'd like to implement the function Music on hold on Opensips. Thank's a lot. Mehdi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Music On Hold Opensips
OpenSIPS is a SIP proxy, not a media endpoint. So, it doesn't do that. On 01/21/2010 12:00 PM, Mehdi Bouchefra wrote: Hello, I'd like to implement the function Music on hold on Opensips. Thank’s a lot. Mehdi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] sched_yield()
Hi,
Now shared memory is 1G (-m 1024), and all memory is dedicated to the virtual
machine (it was shared till now).
But it still happens, just not so often.
I originate the calls for this stress test in Asterisk with the same resources
and looks like Asterisk performs much better than OpenSIPS. How can it be?
In my stress OpenSIPS does no blocking/slow requests. And it's just 4K
concurrent calls, each one is 2-3 min.
Maybe OpenSIPS does too much low level memory management and virtual machine is
not suitable for it (despite that Asterisk runs well over VMware)?
I'm not sure but I have a feeling that 1.4 performed better. What can cause
performance degradation in 1.6? Storing vars on dialog, new malloc()?
gdb) bt
#0 0xb78ad424 in __kernel_vsyscall ()
#1 0xb77e841c in sched_yield () from /lib/i686/cmov/libc.so.6
#2 0x080bf23d in new_avp ()
#3 0x080bf53f in add_avp ()
#4 0x08080e6e in pv_set_avp ()
#5 0x0808229c in pv_set_value ()
#6 0x08053c9d in do_assign ()
#7 0x0805447a in do_action ()
#8 0x08053ebf in run_action_list ()
#9 0x08056e7a in do_action ()
#10 0x08053ebf in run_action_list ()
#11 0x08056e7a in do_action ()
#12 0x08053ebf in run_action_list ()
#13 0x080569d8 in do_action ()
#14 0x08053ebf in run_action_list ()
#15 0x08056e7a in do_action ()
#16 0x08053ebf in run_action_list ()
#17 0x08057d99 in run_top_route ()
#18 0x0808ad6c in receive_msg ()
#19 0x080bd2f2 in udp_rcv_loop ()
#20 0x08069339 in main ()
(gdb) quit
(gdb) bt
#0 0xb78ad424 in __kernel_vsyscall ()
#1 0xb77e841c in sched_yield () from /lib/i686/cmov/libc.so.6
#2 0xb76f52cd in build_cell () from /usr/lib/opensips/modules/tm.so
#3 0xb770ac4a in t_newtran () from /usr/lib/opensips/modules/tm.so
#4 0xb76ff7b8 in t_relay_to () from /usr/lib/opensips/modules/tm.so
#5 0xb770c501 in ?? () from /usr/lib/opensips/modules/tm.so
#6 0x08055030 in do_action ()
#7 0x08053ebf in run_action_list ()
#8 0x08095cf2 in eval_expr ()
#9 0x080958d9 in eval_expr ()
#10 0x08095919 in eval_expr ()
#11 0x080554e2 in do_action ()
#12 0x08053ebf in run_action_list ()
#13 0x080569d8 in do_action ()
#14 0x08053ebf in run_action_list ()
#15 0x08056e7a in do_action ()
#16 0x08053ebf in run_action_list ()
#17 0x08057d99 in run_top_route ()
#18 0x0808ad6c in receive_msg ()
#19 0x080bd2f2 in udp_rcv_loop ()
#20 0x08069339 in main ()
--
Best Regards,
Alex Massover
VoIP RD TL
Jajah Inc.
-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Andrei Dragus
Sent: Thursday, January 21, 2010 3:43 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] sched_yield()
My guess is that there is not enough shared memory. When an allocation
failes OpenSIPS tries to defragment memory to make room which takes a
lot of time and must be done under lock.
Please try to increase the shared memory size and tell me if it
persists.
Alex Massover wrote:
Hi!
Yes, with -DF_MALLOC.
1.6.1 from sources, I build deb package.
I use 128M of shared and 10*1024*1024 private memory (can increase -
no problem).
H, opensipsctl fifo get_statistics all crashes/stops the
opensips.
'fifo uptime' or 'fifo debug' are OK.
strace while 'fifo get_statistics all':
Process 9509 attached - interrupt to quit
pause() = ? ERESTARTNOHAND (To be
restarted)
--- SIGUSR2 (User defined signal 2) @ 0 (0) ---
sigreturn() = ? (mask now [])
pause() = ? ERESTARTNOHAND (To be
restarted)
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
waitpid(-1, [{WIFSIGNALED(s) WTERMSIG(s) == SIGUSR2}], WNOHANG) =
9520
waitpid(-1, 0xbf84b4c8, WNOHANG)= 0
kill(0, SIGTERM)= 0
--- SIGTERM (Terminated) @ 0 (0) ---
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [TERM])
sigreturn() = ? (mask now [])
rt_sigaction(SIGALRM, {0x8065920, [ALRM], SA_RESTART}, {SIG_DFL}, 8)
= 0
alarm(60) = 0
wait4(-1, NULL, 0, NULL)= 9514
wait4(-1, NULL, 0, NULL)= 9519
wait4(-1, NULL, 0, NULL)= 9521
wait4(-1, NULL, 0, NULL)= 9522
wait4(-1, NULL, 0, NULL)= 9512
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
wait4(-1, NULL, 0, NULL)= 9510
wait4(-1, NULL, 0, NULL)= 9516
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
--- SIGCHLD (Child exited) @ 0 (0) ---
sigreturn() = ? (mask now [])
wait4(-1, NULL, 0, NULL)= 9515
wait4(-1, NULL, 0,
Re: [OpenSIPS-Users] Music On Hold Opensips
Hello, An jeu., janv 21, 2010, Alex Balashov schrieb: On 01/21/2010 12:00 PM, Mehdi Bouchefra wrote: I'd like to implement the function Music on hold on Opensips. OpenSIPS is a SIP proxy, not a media endpoint. So, it doesn't do that. If while OpenSIPS routes a call a person presses 'hold' on a telephone, often it will send a INVITE (I think it's called a REINVITE.) What about detecting this 'hold' REINVITE in the route script and redirecting the message to the media server? In the end, another software will have to do the job of the media server (as Alex points out), but it would seem a logical role for OpenSIPS to play in some redirection for phones and their 'hold' buttons. Regards, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Music On Hold Opensips
On Thursday 21 January 2010 17:30:36 opensipsl...@encambio.com wrote: Hello, An jeu., janv 21, 2010, Alex Balashov schrieb: On 01/21/2010 12:00 PM, Mehdi Bouchefra wrote: I'd like to implement the function Music on hold on Opensips. OpenSIPS is a SIP proxy, not a media endpoint. So, it doesn't do that. If while OpenSIPS routes a call a person presses 'hold' on a telephone, often it will send a INVITE (I think it's called a REINVITE.) What about detecting this 'hold' REINVITE in the route script and redirecting the message to the media server? In the end, another software will have to do the job of the media server (as Alex points out), but it would seem a logical role for OpenSIPS to play in some redirection for phones and their 'hold' buttons. No, no, no and no ... OpenSIPS it's a proxy and COULD not do anything in a middle of an stablished dialog (an ongoing call) For doing a MoH server, you will need to use the B2BUA module (and I don't know if could be used for this ..). -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Private IP in registered AOR causing failure
Hello list, An jeu., janv 21, 2010, opensipsl...@encambio.com schrieb: An mer., janv 20, 2010, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: Here's a record I see when I run 'opensipsctl ul show': AOR:: mylogin-osips Contact:: sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm Q=1 Expires:: 560 Callid:: 2b21cdfae784-av13rj1txbsq Cseq:: 2 User-agent:: Bigphone123 Received:: sip:85.182.68.45:2240;transport=TLS State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: tls:80.200.123.45:5061 Methods:: 7999 OpenSIPS is trying to reach the private IP number above from time to time, and I see this in the logs: Jan 19 17:57:20 name.host.tld error opensips[23432]: ERROR:tm:t_uac: attempt to send to 'sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm' failed the problem is not the private IP in the contact, as opensips properly saved the source IP (of the REGISTER) too - see the Received field. So the Received field will be used over the Contact for sending the requests to UAC. Now, what probably goes wrong in your case is that when using TLS/TCP (connection oriented protos), after the REGISTER, the connection is dropped and opensips cannot open later a TCP connection behind a NAT :(By default opensips closes the inactive TCP connections. After running a socket listener on 192.168.0.31 on the OpenSIPS host: $ socat TCP4-LISTEN:2310,bind=192.168.0.31,reuseaddr - SUBSCRIBE sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm SIP/2.0 Via: SIP/2.0/TCP 86.90.39.44;branch=G4z9hb82dK8.f144.0 To: sip:mylogin-os...@name.host.tld;tag=ty6sjh9iz9 From: sip:mylogin-os...@name.host.tld;tag=6c9d4319c74d756e6b696-baa1 CSeq: 11 SUBSCRIBE Call-ID: b1c04118-8...@86.90.39.44 Content-Length: 0 User-Agent: OpenSIPS (1.6.1-tls) Max-Forwards: 70 Event: dialog;sla Contact: sip:prese...@name.host.tld Expires: 610 I'm trying to implement presence by using the presence, presence_xml, pua, and pua_bla modules. So it seems that one of these modules (see event dialog;sla) is getting the contact from the locations table (in AAA on our server) and ignoring the Received header. OpenSIPS replies to messages from UACs such as INVITE and CANCEL properly, and opens connections to the IP in Received. This problem is limited to the SUBSCRIBES sent from one of the presence modules. ...and similar SUBSCRIBE messages (sent from one of the presence modules) are not having this problem. They are almost the same as the one above, but simply don't have a to tag. Greetings, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Music On Hold Opensips
El Jueves, 21 de Enero de 2010, opensipsl...@encambio.com escribió: If while OpenSIPS routes a call a person presses 'hold' on a telephone, often it will send a INVITE (I think it's called a REINVITE.) What about detecting this 'hold' REINVITE in the route script and redirecting the message to the media server? This is annoying as it would break the existing dialog between original endpoints (CSeq broken, SDP broken). Also, the remote endpoint would detect lack of RTP and would abort the call. No, this is not a feasible option at all. Regards. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Music On Hold Opensips
I actually was looking into this last night and found this post from Bogdan http://n2.nabble.com/Request-for-Brain-storming-New-types-of-routes-in-config-tt2693231.html#a2693231 Bogdan mentions in the subject that he is just brainstorming about new types of routing. -- View this message in context: http://n2.nabble.com/Music-On-Hold-Opensips-tp4434891p4435569.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Unsuccessfull upgrade from 1.4.5 to 1.6.1 (RR module)
Hi, I'm running a statefull proxy that in most cases need to relay the calls to a PSTN gateway. After the migration to the Opensips 1.6.1, there is a problem with compatibility / RR module and the gateway (Cisco AS5300). Opensips does not relay 'correctly' (in my case) the ACK messages. The Cisco gateway do not receive ACKs and hangup the call after a timeout. The configuration script is developed on the sipwise template, but it works perfectly in 1.4 version of Opensips. When debugging I see each time more and more headers in ACK packets Record-Route: sip:xx.yy.56.226;lr=on;ftag=4a734ab13a9 Record-Route: sip:xx.yy.56.226;lr=on;ftag=4a734ab13a9 Record-Route: sip:xx.yy.56.226;lr=on;ftag=4a734ab13a9 Record-Route: sip:xx.yy.56.226;lr=on;ftag=4a734ab13a9 Via: SIP/2.0/UDP xx.yy.56.226;branch=z9hG4bKa636.94cf333.2 Via: SIP/2.0/UDP xx.yy.56.226;branch=z9hG4bKa636.94cf333.2 Via: SIP/2.0/UDP xx.yy.56.226;branch=z9hG4bKa636.94cf333.2 Via: SIP/2.0/UDP xx.yy.56.226;branch=z9hG4bKa636.94cf333.2 Via: SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111423e34b43aea64e671405 When disabling the record_route(), and messages go from sip client directly to the gateway - all in ok. When communicating between 2 sip clients on the same proxy - the messages are relayed correctly. What can be the solution? The entire message log: U xx.yy.17.20:5060 - xx.yy.56.226:5060 INVITE sip:987...@xx.yy.56.226 SIP/2.0. Via: SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111426464b44616423ea16ed. From: unknown sip:123...@xx.yy.56.226;tag=152937654c2b. To: sip:987...@xx.yy.56.226. Contact: sip:123...@xx.yy.17.20. Call-ID: D17BF0696DC546A7B436401CD27774720x591c1114. CSeq: 1 INVITE. Max-Forwards: 70. User-Agent: SJphone/1.65.377a (SJ Labs). U xx.yy.56.226:5060 - xx.yy.17.20:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111426464b44616423ea16ed. From: unknown sip:123...@xx.yy.56.226;tag=152937654c2b. To: sip:987...@xx.yy.56.226. Call-ID: D17BF0696DC546A7B436401CD27774720x591c1114. CSeq: 1 INVITE. U xx.yy.56.226:5060 - xx.yy.17.20:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111426464b44616423ea16ed. From: unknown sip:123...@xx.yy.56.226;tag=152937654c2b. To: sip:987...@xx.yy.56.226;tag=c97b4d1cb1f3d0da549e06a8d482ef63.8548. Call-ID: D17BF0696DC546A7B436401CD27774720x591c1114. CSeq: 1 INVITE. U xx.yy.17.20:5060 - xx.yy.56.226:5060 ACK sip:987...@xx.yy.56.226 SIP/2.0. Via: SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111426464b44616423ea16ed. From: unknown sip:123...@xx.yy.56.226;tag=152937654c2b. To: sip:987...@xx.yy.56.226;tag=c97b4d1cb1f3d0da549e06a8d482ef63.8548. Call-ID: D17BF0696DC546A7B436401CD27774720x591c1114. CSeq: 1 ACK. Max-Forwards: 70. User-Agent: SJphone/1.65.377a (SJ Labs). Content-Length: 0. U xx.yy.17.20:5060 - xx.yy.56.226:5060 INVITE sip:987...@xx.yy.56.226 SIP/2.0. Via: SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111426474b446164164716f0. From: unknown sip:123...@xx.yy.56.226;tag=152937654c2b. To: sip:987...@xx.yy.56.226. Contact: sip:123...@xx.yy.17.20. Call-ID: D17BF0696DC546A7B436401CD27774720x591c1114. CSeq: 2 INVITE. Max-Forwards: 70. User-Agent: SJphone/1.65.377a (SJ Labs). Content-Length: 362. Content-Type: application/sdp. Supported: replaces,norefersub,timer. Proxy-Authorization: Digest username=123456,realm=xx.yy.56.226,nonce=4b43ec22000883661dbea2617b91d28401983dd85b7d,uri=sip:987...@xx.yy.56.226 ,response=91a62cf62af7870c1e24d6c35857e78d. U xx.yy.56.226:5060 - xx.yy.17.20:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111426474b446164164716f0. From: unknown sip:123...@xx.yy.56.226;tag=152937654c2b. To: sip:987...@xx.yy.56.226. Call-ID: D17BF0696DC546A7B436401CD27774720x591c1114. CSeq: 2 INVITE. Server: OpenSIPS (1.6.1-notls (x86_64/linux)). Content-Length: 0. U xx.yy.56.226:5060 - xx.yy.17.20:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111426474b446164164716f0. From: unknown sip:123...@xx.yy.56.226;tag=152937654c2b. To: sip:987...@xx.yy.56.226. Call-ID: D17BF0696DC546A7B436401CD27774720x591c1114. CSeq: 2 INVITE. U xx.yy.56.226:5060 - xx.yy.25.114:5060 INVITE sip:987...@xx.yy.25.114:5060;transport=udp SIP/2.0. Record-Route: sip:xx.yy.56.226;lr=on;ftag=152937654c2b;did=b39.e643f344. Via: SIP/2.0/UDP xx.yy.56.226;branch=z9hG4bKf69b.0ced9bb7.0. Via: SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111426474b446164164716f0. From: unknown sip:123...@xx.yy.56.226;tag=152937654c2b. To: sip:987...@xx.yy.56.226. Contact: sip:123...@xx.yy.17.20. Call-ID: D17BF0696DC546A7B436401CD27774720x591c1114. CSeq: 2 INVITE. Max-Forwards: 69. U xx.yy.25.114:5060 - xx.yy.56.226:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP xx.yy.56.226;branch=z9hG4bKf69b.0ced9bb7.0,SIP/2.0/UDP xx.yy.17.20;branch=z9hG4bK591c111426474b446164164716f0. From: unknown
[OpenSIPS-Users] Query regarding Rtp Proxy opensips
Hi Everyone,
I'm trying to record calls using rtpproxy. i called call_recording() while
i get invite message and on onreply route.
as follows:
I) if (is_method(INVITE)){
force_rtp_proxy();
start_recording();
2) onreply_route[1] {
if ((isflagset(5) || isbflagset(6)) status=~(183)|(2[0-9][0-9])) {
force_rtp_proxy();
start_recording();
}
When i make a call between my sip users a rtp files are storing in src
directory. I'm confused how to decode and play these files. For each call it
is saving two files
11784d24-9142b...@192.168.3.10=73818ee84fc2bab5o3;1.a.rtp
11784d24-9142b...@192.168.3.10=73818ee84fc2bab5o3;1.o.rtp.
I Read some where that .a.rtp is stream from caller, while .o.rtp is
stream from callee.
how to decode these files and listen wheter the call is recorded or not?
Thanks in Advance.
--
View this message in context:
http://n2.nabble.com/Query-regarding-Rtp-Proxy-opensips-tp4438620p4438620.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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