Re: [OpenSIPS-Users] Possible Drouting bug
Yes I tried that but still same issue. I call goes_to_gw and after that
attributes are filled.
On Tue, Mar 28, 2017 at 3:25 PM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:
> Hello John,
>
> Do you use partitions ? Is the use_partition enabled ? If not, the
> use_next_gw() should be used like:
> use_next_gw( "$avp(rules_attributes)","$avp(gw_attributes)")
>
> (the first param, the partition, is not to be provided)
>
> Could you check if this solves the problem ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 03/23/2017 09:19 AM, John Nash wrote:
>
> I am using drouting and recently tried to use gateway attribute. I call
> ...
>
> do_routing("$avp(int_grp_id)","WF","$avp(gw_whitelist)" ,
> "$avp(rules_attributes)","$avp(gw_attributes)"))
>
> After this call I can see $avp(gw_attributes) is populated frp, attr
> column of dr_gateways table.
>
> but when i call following ...
> use_next_gw(,"$avp(rules_attributes)","$avp(gw_attributes)")
>
> $avp(gw_attributes) becomes empty
>
>
> If i call next_routing() instead of use_next_gw then $avp(gw_attributes)
> retains old value but does not populate new value
>
>
>
>
>
>
> ___
> Users mailing
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
___
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[OpenSIPS-Users] Possible Drouting bug
I am using drouting and recently tried to use gateway attribute. I call ...
do_routing("$avp(int_grp_id)","WF","$avp(gw_whitelist)" ,
"$avp(rules_attributes)","$avp(gw_attributes)"))
After this call I can see $avp(gw_attributes) is populated frp, attr column
of dr_gateways table.
but when i call following ...
use_next_gw(,"$avp(rules_attributes)","$avp(gw_attributes)")
$avp(gw_attributes) becomes empty
If i call next_routing() instead of use_next_gw then $avp(gw_attributes)
retains old value but does not populate new value
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[OpenSIPS-Users] Store Connection address from SDP to a variable
I can get C line using {sdp.line} but how can I separate "Connection
address" and store in some variable? csv transformation is a cool option
but that requires coma separated string.
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Re: [OpenSIPS-Users] Quest to find memory leak
Yes this patch solves my problem. Thank you. I wonder why it did not show as memory leak in dumps. On Fri, Mar 10, 2017 at 3:19 PM, Răzvan Crainea <raz...@opensips.org> wrote: > Hi, John! > > I think you might spot something. Can you apply this patch[1] and try > again? > > [1] https://gist.github.com/razvancrainea/03a43bfa8b554a7ca89f2740a3c54c96 > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/10/2017 10:02 AM, John Nash wrote: > > Dear Razvan, > > I think I found one issue in do_routing function if i pass gw_whitelist > then only i see this drop of memory. If I skip this parameter private > memory does not increase with every call. > > Regards > > Manoj > > On Thu, Mar 9, 2017 at 7:29 PM, John Nash <john.nash...@gmail.com> wrote: > >> If I use 2.2 will it give clearer picture of memory allocation? or 2.3 >> >> On Thu, Mar 9, 2017 at 7:21 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> There's no need for that. No function should leak in any circumstances >>> :) >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/09/2017 03:27 PM, John Nash wrote: >>> >>> OK..May i send you my script privately? >>> >>> On Thu, Mar 9, 2017 at 6:13 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> Hi, John! >>>> >>>> No, I nothing is suspicious. Definitely not from the drouting module. >>>> Try to make two captures: one after 10 calls, another one after 20 >>>> calls. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 03/09/2017 01:50 PM, John Nash wrote: >>>> >>>> Do you see anything suspicious in the latest mem dump? >>>> >>>> On Wed, Mar 8, 2017 at 7:20 PM, John Nash <john.nash...@gmail.com> >>>> wrote: >>>> >>>>> One more useful info. I disabled drouting functions and just rewrote >>>>> RURI to hardcoded address keeping rest of the functions same and I do not >>>>> see drop in private memory of that process. >>>>> >>>>> On Wed, Mar 8, 2017 at 4:40 PM, John Nash <john.nash...@gmail.com> >>>>> wrote: >>>>> >>>>>> OK Here is the dump >>>>>> https://drive.google.com/open?id=0BxJKNwFalcRMX0xDUlRIa2VUdG8 >>>>>> >>>>>> >>>>>> I increased syslog message rate to 50, Made around 10 call >>>>>> attempts. Waited for some time and made sure no call is on server and >>>>>> then >>>>>> sent signal to dump memory to the process ID i suspect. >>>>>> >>>>>> On Wed, Mar 8, 2017 at 4:07 PM, Răzvan Crainea <raz...@opensips.org> >>>>>> wrote: >>>>>> >>>>>>> No, you should not kill any process. Simply send a SIGUSR1 to the >>>>>>> process you suspect. >>>>>>> >>>>>>> Răzvan Crainea >>>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>>> >>>>>>> On 03/08/2017 12:28 PM, John Nash wrote: >>>>>>> >>>>>>> Sorry...Should I kill only the process where i see memory leak? >>>>>>> >>>>>>> >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > 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] Quest to find memory leak
Dear Razvan, I think I found one issue in do_routing function if i pass gw_whitelist then only i see this drop of memory. If I skip this parameter private memory does not increase with every call. Regards Manoj On Thu, Mar 9, 2017 at 7:29 PM, John Nash <john.nash...@gmail.com> wrote: > If I use 2.2 will it give clearer picture of memory allocation? or 2.3 > > On Thu, Mar 9, 2017 at 7:21 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> There's no need for that. No function should leak in any circumstances :) >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/09/2017 03:27 PM, John Nash wrote: >> >> OK..May i send you my script privately? >> >> On Thu, Mar 9, 2017 at 6:13 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> Hi, John! >>> >>> No, I nothing is suspicious. Definitely not from the drouting module. >>> Try to make two captures: one after 10 calls, another one after 20 calls. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/09/2017 01:50 PM, John Nash wrote: >>> >>> Do you see anything suspicious in the latest mem dump? >>> >>> On Wed, Mar 8, 2017 at 7:20 PM, John Nash <john.nash...@gmail.com> >>> wrote: >>> >>>> One more useful info. I disabled drouting functions and just rewrote >>>> RURI to hardcoded address keeping rest of the functions same and I do not >>>> see drop in private memory of that process. >>>> >>>> On Wed, Mar 8, 2017 at 4:40 PM, John Nash <john.nash...@gmail.com> >>>> wrote: >>>> >>>>> OK Here is the dump >>>>> https://drive.google.com/open?id=0BxJKNwFalcRMX0xDUlRIa2VUdG8 >>>>> >>>>> >>>>> I increased syslog message rate to 50, Made around 10 call >>>>> attempts. Waited for some time and made sure no call is on server and then >>>>> sent signal to dump memory to the process ID i suspect. >>>>> >>>>> On Wed, Mar 8, 2017 at 4:07 PM, Răzvan Crainea <raz...@opensips.org> >>>>> wrote: >>>>> >>>>>> No, you should not kill any process. Simply send a SIGUSR1 to the >>>>>> process you suspect. >>>>>> >>>>>> Răzvan Crainea >>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>> >>>>>> On 03/08/2017 12:28 PM, John Nash wrote: >>>>>> >>>>>> Sorry...Should I kill only the process where i see memory leak? >>>>>> >>>>>> >> ___ >> 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] Quest to find memory leak
If I use 2.2 will it give clearer picture of memory allocation? or 2.3 On Thu, Mar 9, 2017 at 7:21 PM, Răzvan Crainea <raz...@opensips.org> wrote: > There's no need for that. No function should leak in any circumstances :) > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/09/2017 03:27 PM, John Nash wrote: > > OK..May i send you my script privately? > > On Thu, Mar 9, 2017 at 6:13 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> Hi, John! >> >> No, I nothing is suspicious. Definitely not from the drouting module. >> Try to make two captures: one after 10 calls, another one after 20 calls. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/09/2017 01:50 PM, John Nash wrote: >> >> Do you see anything suspicious in the latest mem dump? >> >> On Wed, Mar 8, 2017 at 7:20 PM, John Nash <john.nash...@gmail.com> wrote: >> >>> One more useful info. I disabled drouting functions and just rewrote >>> RURI to hardcoded address keeping rest of the functions same and I do not >>> see drop in private memory of that process. >>> >>> On Wed, Mar 8, 2017 at 4:40 PM, John Nash <john.nash...@gmail.com> >>> wrote: >>> >>>> OK Here is the dump >>>> https://drive.google.com/open?id=0BxJKNwFalcRMX0xDUlRIa2VUdG8 >>>> >>>> >>>> I increased syslog message rate to 50, Made around 10 call >>>> attempts. Waited for some time and made sure no call is on server and then >>>> sent signal to dump memory to the process ID i suspect. >>>> >>>> On Wed, Mar 8, 2017 at 4:07 PM, Răzvan Crainea <raz...@opensips.org> >>>> wrote: >>>> >>>>> No, you should not kill any process. Simply send a SIGUSR1 to the >>>>> process you suspect. >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>> >>>>> On 03/08/2017 12:28 PM, John Nash wrote: >>>>> >>>>> Sorry...Should I kill only the process where i see memory leak? >>>>> >>>>> > ___ > 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] Quest to find memory leak
OK..May i send you my script privately? On Thu, Mar 9, 2017 at 6:13 PM, Răzvan Crainea <raz...@opensips.org> wrote: > Hi, John! > > No, I nothing is suspicious. Definitely not from the drouting module. > Try to make two captures: one after 10 calls, another one after 20 calls. > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/09/2017 01:50 PM, John Nash wrote: > > Do you see anything suspicious in the latest mem dump? > > On Wed, Mar 8, 2017 at 7:20 PM, John Nash <john.nash...@gmail.com> wrote: > >> One more useful info. I disabled drouting functions and just rewrote RURI >> to hardcoded address keeping rest of the functions same and I do not see >> drop in private memory of that process. >> >> On Wed, Mar 8, 2017 at 4:40 PM, John Nash <john.nash...@gmail.com> wrote: >> >>> OK Here is the dump >>> https://drive.google.com/open?id=0BxJKNwFalcRMX0xDUlRIa2VUdG8 >>> >>> >>> I increased syslog message rate to 50, Made around 10 call attempts. >>> Waited for some time and made sure no call is on server and then sent >>> signal to dump memory to the process ID i suspect. >>> >>> On Wed, Mar 8, 2017 at 4:07 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> No, you should not kill any process. Simply send a SIGUSR1 to the >>>> process you suspect. >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 03/08/2017 12:28 PM, John Nash wrote: >>>> >>>> Sorry...Should I kill only the process where i see memory leak? >>>> >>>> On Wed, Mar 8, 2017 at 3:41 PM, Răzvan Crainea <raz...@opensips.org> >>>> wrote: >>>> >>>>> use only memdump set to 1. >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>> >>>>> On 03/08/2017 12:11 PM, John Nash wrote: >>>>> >>>>> Ok i will give another try what should be the values of memdump and >>>>> memlog >>>>> >>>>> On Wed, Mar 8, 2017 at 3:13 PM, Răzvan Crainea <raz...@opensips.org> >>>>> wrote: >>>>> >>>>>> Hi, John! >>>>>> >>>>>> The traces you showed me are incomplete: they do not have all the >>>>>> memory chunks allocated, thus I can't say wether something is wrong or >>>>>> not. >>>>>> As I said earlier, it is normal for opensips to use extra memory >>>>>> every call. But after a while, this should stabilize. After a while might >>>>>> mean more than 1000k calls. As long as you never reach the upper limit of >>>>>> the memory, you can't conclude that there is a memory leak. Even then, >>>>>> you're limit might be too low for the kind of traffic you are doing, so >>>>>> it >>>>>> still might not be a memory leak. But only then it is worth to >>>>>> investigate. >>>>>> When we investigate, we need all the data (i.e. the entire trace of >>>>>> the memory dump). >>>>>> So please try to send as many calls as possilble, and if this issue >>>>>> still persists, make a pkg memory dump when the server is in idle mode >>>>>> and >>>>>> send it over. >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Răzvan Crainea >>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>> >>>>>> On 03/08/2017 11:26 AM, John Nash wrote: >>>>>> >>>>>> any suggestion for me?..should i try to crash opensips by sending >>>>>> many calls? >>>>>> >>>>>> On Tue, Mar 7, 2017 at 4:54 PM, John Nash <john.nash...@gmail.com> >>>>>> wrote: >>>>>> >>>>>>> version: opensips 2.1.5 (x86_64/linux) >>>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >>>>>>> DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN >>>>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>>>>>>
Re: [OpenSIPS-Users] Quest to find memory leak
Do you see anything suspicious in the latest mem dump? On Wed, Mar 8, 2017 at 7:20 PM, John Nash <john.nash...@gmail.com> wrote: > One more useful info. I disabled drouting functions and just rewrote RURI > to hardcoded address keeping rest of the functions same and I do not see > drop in private memory of that process. > > On Wed, Mar 8, 2017 at 4:40 PM, John Nash <john.nash...@gmail.com> wrote: > >> OK Here is the dump >> https://drive.google.com/open?id=0BxJKNwFalcRMX0xDUlRIa2VUdG8 >> >> >> I increased syslog message rate to 50, Made around 10 call attempts. >> Waited for some time and made sure no call is on server and then sent >> signal to dump memory to the process ID i suspect. >> >> On Wed, Mar 8, 2017 at 4:07 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> No, you should not kill any process. Simply send a SIGUSR1 to the >>> process you suspect. >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/08/2017 12:28 PM, John Nash wrote: >>> >>> Sorry...Should I kill only the process where i see memory leak? >>> >>> On Wed, Mar 8, 2017 at 3:41 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> use only memdump set to 1. >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 03/08/2017 12:11 PM, John Nash wrote: >>>> >>>> Ok i will give another try what should be the values of memdump and >>>> memlog >>>> >>>> On Wed, Mar 8, 2017 at 3:13 PM, Răzvan Crainea <raz...@opensips.org> >>>> wrote: >>>> >>>>> Hi, John! >>>>> >>>>> The traces you showed me are incomplete: they do not have all the >>>>> memory chunks allocated, thus I can't say wether something is wrong or >>>>> not. >>>>> As I said earlier, it is normal for opensips to use extra memory every >>>>> call. But after a while, this should stabilize. After a while might mean >>>>> more than 1000k calls. As long as you never reach the upper limit of the >>>>> memory, you can't conclude that there is a memory leak. Even then, you're >>>>> limit might be too low for the kind of traffic you are doing, so it still >>>>> might not be a memory leak. But only then it is worth to investigate. >>>>> When we investigate, we need all the data (i.e. the entire trace of >>>>> the memory dump). >>>>> So please try to send as many calls as possilble, and if this issue >>>>> still persists, make a pkg memory dump when the server is in idle mode and >>>>> send it over. >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>> >>>>> On 03/08/2017 11:26 AM, John Nash wrote: >>>>> >>>>> any suggestion for me?..should i try to crash opensips by sending many >>>>> calls? >>>>> >>>>> On Tue, Mar 7, 2017 at 4:54 PM, John Nash <john.nash...@gmail.com> >>>>> wrote: >>>>> >>>>>> version: opensips 2.1.5 (x86_64/linux) >>>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >>>>>> DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>>>>> MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>>>>> git revision: 39b19dd >>>>>> main.c compiled on 19:27:59 Mar 5 2017 with gcc 4.4.7 >>>>>> >>>>>> memory stabilizing in time? Or it is continously decreasing? >>>>>> Yes, that's how you should make the dump. >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Răzvan Crainea >>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>> >>>>>> >>>>>> >>>>> ___ >>>>> Users mailing list >>>>> Users@lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>>> >>>> ___ >>>> Users mailing >>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> ___ Users mailing list >>>> Users@lists.opensips.org http://lists.opensips.org/cgi- >>>> bin/mailman/listinfo/users >>> >>> ___ >>> Users mailing >>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> ___ >>> 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] Quest to find memory leak
One more useful info. I disabled drouting functions and just rewrote RURI to hardcoded address keeping rest of the functions same and I do not see drop in private memory of that process. On Wed, Mar 8, 2017 at 4:40 PM, John Nash <john.nash...@gmail.com> wrote: > OK Here is the dump > https://drive.google.com/open?id=0BxJKNwFalcRMX0xDUlRIa2VUdG8 > > > I increased syslog message rate to 50, Made around 10 call attempts. > Waited for some time and made sure no call is on server and then sent > signal to dump memory to the process ID i suspect. > > On Wed, Mar 8, 2017 at 4:07 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> No, you should not kill any process. Simply send a SIGUSR1 to the process >> you suspect. >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/08/2017 12:28 PM, John Nash wrote: >> >> Sorry...Should I kill only the process where i see memory leak? >> >> On Wed, Mar 8, 2017 at 3:41 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> use only memdump set to 1. >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/08/2017 12:11 PM, John Nash wrote: >>> >>> Ok i will give another try what should be the values of memdump and >>> memlog >>> >>> On Wed, Mar 8, 2017 at 3:13 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> Hi, John! >>>> >>>> The traces you showed me are incomplete: they do not have all the >>>> memory chunks allocated, thus I can't say wether something is wrong or not. >>>> As I said earlier, it is normal for opensips to use extra memory every >>>> call. But after a while, this should stabilize. After a while might mean >>>> more than 1000k calls. As long as you never reach the upper limit of the >>>> memory, you can't conclude that there is a memory leak. Even then, you're >>>> limit might be too low for the kind of traffic you are doing, so it still >>>> might not be a memory leak. But only then it is worth to investigate. >>>> When we investigate, we need all the data (i.e. the entire trace of the >>>> memory dump). >>>> So please try to send as many calls as possilble, and if this issue >>>> still persists, make a pkg memory dump when the server is in idle mode and >>>> send it over. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 03/08/2017 11:26 AM, John Nash wrote: >>>> >>>> any suggestion for me?..should i try to crash opensips by sending many >>>> calls? >>>> >>>> On Tue, Mar 7, 2017 at 4:54 PM, John Nash <john.nash...@gmail.com> >>>> wrote: >>>> >>>>> version: opensips 2.1.5 (x86_64/linux) >>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >>>>> DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>>>> MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>>>> git revision: 39b19dd >>>>> main.c compiled on 19:27:59 Mar 5 2017 with gcc 4.4.7 >>>>> >>>>> memory stabilizing in time? Or it is continously decreasing? >>>>> Yes, that's how you should make the dump. >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>> >>>>> >>>>> >>>> ___ >>>> Users mailing list >>>> Users@lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> ___ >>> Users mailing >>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> ___ Users mailing list >>> Users@lists.opensips.org http://lists.opensips.org/cgi- >>> bin/mailman/listinfo/users >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> ___ >> 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] Quest to find memory leak
OK Here is the dump https://drive.google.com/open?id=0BxJKNwFalcRMX0xDUlRIa2VUdG8 I increased syslog message rate to 50, Made around 10 call attempts. Waited for some time and made sure no call is on server and then sent signal to dump memory to the process ID i suspect. On Wed, Mar 8, 2017 at 4:07 PM, Răzvan Crainea <raz...@opensips.org> wrote: > No, you should not kill any process. Simply send a SIGUSR1 to the process > you suspect. > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/08/2017 12:28 PM, John Nash wrote: > > Sorry...Should I kill only the process where i see memory leak? > > On Wed, Mar 8, 2017 at 3:41 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> use only memdump set to 1. >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/08/2017 12:11 PM, John Nash wrote: >> >> Ok i will give another try what should be the values of memdump and memlog >> >> On Wed, Mar 8, 2017 at 3:13 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> Hi, John! >>> >>> The traces you showed me are incomplete: they do not have all the memory >>> chunks allocated, thus I can't say wether something is wrong or not. >>> As I said earlier, it is normal for opensips to use extra memory every >>> call. But after a while, this should stabilize. After a while might mean >>> more than 1000k calls. As long as you never reach the upper limit of the >>> memory, you can't conclude that there is a memory leak. Even then, you're >>> limit might be too low for the kind of traffic you are doing, so it still >>> might not be a memory leak. But only then it is worth to investigate. >>> When we investigate, we need all the data (i.e. the entire trace of the >>> memory dump). >>> So please try to send as many calls as possilble, and if this issue >>> still persists, make a pkg memory dump when the server is in idle mode and >>> send it over. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/08/2017 11:26 AM, John Nash wrote: >>> >>> any suggestion for me?..should i try to crash opensips by sending many >>> calls? >>> >>> On Tue, Mar 7, 2017 at 4:54 PM, John Nash <john.nash...@gmail.com> >>> wrote: >>> >>>> version: opensips 2.1.5 (x86_64/linux) >>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >>>> DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>>> MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>>> git revision: 39b19dd >>>> main.c compiled on 19:27:59 Mar 5 2017 with gcc 4.4.7 >>>> >>>> memory stabilizing in time? Or it is continously decreasing? >>>> Yes, that's how you should make the dump. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> >>>> >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> ___ Users mailing list >> Users@lists.opensips.org http://lists.opensips.org/cgi- >> bin/mailman/listinfo/users > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ___ > 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] Quest to find memory leak
Sorry...Should I kill only the process where i see memory leak? On Wed, Mar 8, 2017 at 3:41 PM, Răzvan Crainea <raz...@opensips.org> wrote: > use only memdump set to 1. > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/08/2017 12:11 PM, John Nash wrote: > > Ok i will give another try what should be the values of memdump and memlog > > On Wed, Mar 8, 2017 at 3:13 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> Hi, John! >> >> The traces you showed me are incomplete: they do not have all the memory >> chunks allocated, thus I can't say wether something is wrong or not. >> As I said earlier, it is normal for opensips to use extra memory every >> call. But after a while, this should stabilize. After a while might mean >> more than 1000k calls. As long as you never reach the upper limit of the >> memory, you can't conclude that there is a memory leak. Even then, you're >> limit might be too low for the kind of traffic you are doing, so it still >> might not be a memory leak. But only then it is worth to investigate. >> When we investigate, we need all the data (i.e. the entire trace of the >> memory dump). >> So please try to send as many calls as possilble, and if this issue still >> persists, make a pkg memory dump when the server is in idle mode and send >> it over. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/08/2017 11:26 AM, John Nash wrote: >> >> any suggestion for me?..should i try to crash opensips by sending many >> calls? >> >> On Tue, Mar 7, 2017 at 4:54 PM, John Nash <john.nash...@gmail.com> wrote: >> >>> version: opensips 2.1.5 (x86_64/linux) >>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >>> DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>> MAX_URI_SIZE 1024, BUF_SIZE 65535 >>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>> git revision: 39b19dd >>> main.c compiled on 19:27:59 Mar 5 2017 with gcc 4.4.7 >>> >>> memory stabilizing in time? Or it is continously decreasing? >>> Yes, that's how you should make the dump. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> >>> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > 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] Quest to find memory leak
Ok i will give another try what should be the values of memdump and memlog On Wed, Mar 8, 2017 at 3:13 PM, Răzvan Crainea <raz...@opensips.org> wrote: > Hi, John! > > The traces you showed me are incomplete: they do not have all the memory > chunks allocated, thus I can't say wether something is wrong or not. > As I said earlier, it is normal for opensips to use extra memory every > call. But after a while, this should stabilize. After a while might mean > more than 1000k calls. As long as you never reach the upper limit of the > memory, you can't conclude that there is a memory leak. Even then, you're > limit might be too low for the kind of traffic you are doing, so it still > might not be a memory leak. But only then it is worth to investigate. > When we investigate, we need all the data (i.e. the entire trace of the > memory dump). > So please try to send as many calls as possilble, and if this issue still > persists, make a pkg memory dump when the server is in idle mode and send > it over. > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/08/2017 11:26 AM, John Nash wrote: > > any suggestion for me?..should i try to crash opensips by sending many > calls? > > On Tue, Mar 7, 2017 at 4:54 PM, John Nash <john.nash...@gmail.com> wrote: > >> version: opensips 2.1.5 (x86_64/linux) >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >> DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> git revision: 39b19dd >> main.c compiled on 19:27:59 Mar 5 2017 with gcc 4.4.7 >> >> memory stabilizing in time? Or it is continously decreasing? >> Yes, that's how you should make the dump. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.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] Quest to find memory leak
any suggestion for me?..should i try to crash opensips by sending many calls? On Tue, Mar 7, 2017 at 4:54 PM, John Nash <john.nash...@gmail.com> wrote: > version: opensips 2.1.5 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > git revision: 39b19dd > main.c compiled on 19:27:59 Mar 5 2017 with gcc 4.4.7 > > > On Tue, Mar 7, 2017 at 4:25 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> What allocator are you using? Can you post the output of 'opensips -V'? >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/07/2017 12:23 PM, John Nash wrote: >> >> Please note when i call do_routing in such a way that its unable to find >> any rules matching and reject call i do not see free memory drop. But if it >> finds a route, sends call to that gateway memory drops with each attempt. >> >> On Tue, Mar 7, 2017 at 3:17 PM, John Nash <john.nash...@gmail.com> wrote: >> >>> only 6 or 7 calls >>> >>> On Tue, Mar 7, 2017 at 3:09 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> So I understand that after ~3K calls, that process completely runs out >>>> of memory? >>>> How many calls have you done before this trace: >>>> http://pastebin.com/9Ge2NEVQ >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 03/07/2017 11:32 AM, John Nash wrote: >>>> >>>> when I check stats after a call attempt pkmem:7-free_size:: 3304280 >>>> >>>> In this entry with every call I see a drop of 1000 bytes around and >>>> this never restores. >>>> >>>> On Tue, Mar 7, 2017 at 2:16 PM, Răzvan Crainea <raz...@opensips.org> >>>> wrote: >>>> >>>>> Hi, John! >>>>> >>>>> Again, this trace doesn't show any leak. >>>>> Are you sure you are having a private memory leak and not a shared >>>>> memory leak? >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>> >>>>> On 03/06/2017 08:09 PM, John Nash wrote: >>>>> >>>>> here is another trace >>>>> http://pastebin.com/9Ge2NEVQ >>>>> >>>>> I see lot of alloc request but no free. >>>>> >>>>> On Mon, Mar 6, 2017 at 6:57 PM, John Nash <john.nash...@gmail.com> >>>>> wrote: >>>>> >>>>>> Ok will try that. Is it possible that wrong usage of drouting may >>>>>> cause this to happen instead of actual leak?... What are the things >>>>>> private >>>>>> memory is used for? >>>>>> >>>>>> On Mon, Mar 6, 2017 at 6:48 PM, Răzvan Crainea <raz...@opensips.org> >>>>>> wrote: >>>>>> >>>>>>> Hi, John! >>>>>>> >>>>>>> From the dump you sent, I don't see any leaks. Perhaps some of those >>>>>>> fragments increase over time. Can you make a memory dump after the >>>>>>> server >>>>>>> runs some time, like after it gets 100 messages? >>>>>>> >>>>>>> Best regards, >>>>>>> >>>>>>> Răzvan Crainea >>>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>>> >>>>>>> On 03/06/2017 03:02 PM, John Nash wrote: >>>>>>> >>>>>>> Here is the dump >>>>>>> http://pastebin.com/DTEHF5Vc >>>>>>> >>>>>>> On Mon, Mar 6, 2017 at 6:20 PM, Răzvan Crainea <raz...@opensips.org> >>>>>>> wrote: >>>>>>> >>>>>>>> None of the "actions" you are talking about have big impact on >>>>>>>> private memory, but the shared one. Better do the dump and send it >>>>>>>> over
[OpenSIPS-Users] onreply_route question
I am using t_on_failure("external_failure"); t_on_reply("external_reply");
before calling do_routing function. I expected failure replies to go
to failure_route[external_failure] only but failure replies also going
to onreply_route[external_reply] along with failure_route[external_failure]
Is there something wrong I am doing?
___
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Users@lists.opensips.org
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Re: [OpenSIPS-Users] Compiler flags disabled
please ignore i had selected wrong option to clean. On Tue, Mar 7, 2017 at 9:09 PM, John Nash <john.nash...@gmail.com> wrote: > I was trying to compile on fresh linux cent OS. But when i run make > menuconfig i find that I am not able to see "Configure Compile Flags " > seems disabled. Other options I can go inside like exclude modules etc > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Compiler flags disabled
I was trying to compile on fresh linux cent OS. But when i run make menuconfig i find that I am not able to see "Configure Compile Flags " seems disabled. Other options I can go inside like exclude modules etc ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Quest to find memory leak
version: opensips 2.1.5 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: 39b19dd main.c compiled on 19:27:59 Mar 5 2017 with gcc 4.4.7 On Tue, Mar 7, 2017 at 4:25 PM, Răzvan Crainea <raz...@opensips.org> wrote: > What allocator are you using? Can you post the output of 'opensips -V'? > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/07/2017 12:23 PM, John Nash wrote: > > Please note when i call do_routing in such a way that its unable to find > any rules matching and reject call i do not see free memory drop. But if it > finds a route, sends call to that gateway memory drops with each attempt. > > On Tue, Mar 7, 2017 at 3:17 PM, John Nash <john.nash...@gmail.com> wrote: > >> only 6 or 7 calls >> >> On Tue, Mar 7, 2017 at 3:09 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> So I understand that after ~3K calls, that process completely runs out >>> of memory? >>> How many calls have you done before this trace: >>> http://pastebin.com/9Ge2NEVQ >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/07/2017 11:32 AM, John Nash wrote: >>> >>> when I check stats after a call attempt pkmem:7-free_size:: 3304280 >>> >>> In this entry with every call I see a drop of 1000 bytes around and this >>> never restores. >>> >>> On Tue, Mar 7, 2017 at 2:16 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> Hi, John! >>>> >>>> Again, this trace doesn't show any leak. >>>> Are you sure you are having a private memory leak and not a shared >>>> memory leak? >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 03/06/2017 08:09 PM, John Nash wrote: >>>> >>>> here is another trace >>>> http://pastebin.com/9Ge2NEVQ >>>> >>>> I see lot of alloc request but no free. >>>> >>>> On Mon, Mar 6, 2017 at 6:57 PM, John Nash <john.nash...@gmail.com> >>>> wrote: >>>> >>>>> Ok will try that. Is it possible that wrong usage of drouting may >>>>> cause this to happen instead of actual leak?... What are the things >>>>> private >>>>> memory is used for? >>>>> >>>>> On Mon, Mar 6, 2017 at 6:48 PM, Răzvan Crainea <raz...@opensips.org> >>>>> wrote: >>>>> >>>>>> Hi, John! >>>>>> >>>>>> From the dump you sent, I don't see any leaks. Perhaps some of those >>>>>> fragments increase over time. Can you make a memory dump after the server >>>>>> runs some time, like after it gets 100 messages? >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Răzvan Crainea >>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>> >>>>>> On 03/06/2017 03:02 PM, John Nash wrote: >>>>>> >>>>>> Here is the dump >>>>>> http://pastebin.com/DTEHF5Vc >>>>>> >>>>>> On Mon, Mar 6, 2017 at 6:20 PM, Răzvan Crainea <raz...@opensips.org> >>>>>> wrote: >>>>>> >>>>>>> None of the "actions" you are talking about have big impact on >>>>>>> private memory, but the shared one. Better do the dump and send it over >>>>>>> to >>>>>>> point out what is "eating" memory. >>>>>>> >>>>>>> Best regards, >>>>>>> >>>>>>> Răzvan Crainea >>>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>>> >>>>>>> On 03/06/2017 02:39 PM, John Nash wrote: >>>>>>> >>>>>>> with every call attempt it decreases. I tried some changes by >>>>>>> rejecting invite before drouting call (That means after auth , >>>>>>> dispatch
Re: [OpenSIPS-Users] Quest to find memory leak
Please note when i call do_routing in such a way that its unable to find any rules matching and reject call i do not see free memory drop. But if it finds a route, sends call to that gateway memory drops with each attempt. On Tue, Mar 7, 2017 at 3:17 PM, John Nash <john.nash...@gmail.com> wrote: > only 6 or 7 calls > > On Tue, Mar 7, 2017 at 3:09 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> So I understand that after ~3K calls, that process completely runs out of >> memory? >> How many calls have you done before this trace: >> http://pastebin.com/9Ge2NEVQ >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/07/2017 11:32 AM, John Nash wrote: >> >> when I check stats after a call attempt pkmem:7-free_size:: 3304280 >> >> In this entry with every call I see a drop of 1000 bytes around and this >> never restores. >> >> On Tue, Mar 7, 2017 at 2:16 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> Hi, John! >>> >>> Again, this trace doesn't show any leak. >>> Are you sure you are having a private memory leak and not a shared >>> memory leak? >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/06/2017 08:09 PM, John Nash wrote: >>> >>> here is another trace >>> http://pastebin.com/9Ge2NEVQ >>> >>> I see lot of alloc request but no free. >>> >>> On Mon, Mar 6, 2017 at 6:57 PM, John Nash <john.nash...@gmail.com> >>> wrote: >>> >>>> Ok will try that. Is it possible that wrong usage of drouting may cause >>>> this to happen instead of actual leak?... What are the things private >>>> memory is used for? >>>> >>>> On Mon, Mar 6, 2017 at 6:48 PM, Răzvan Crainea <raz...@opensips.org> >>>> wrote: >>>> >>>>> Hi, John! >>>>> >>>>> From the dump you sent, I don't see any leaks. Perhaps some of those >>>>> fragments increase over time. Can you make a memory dump after the server >>>>> runs some time, like after it gets 100 messages? >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>> >>>>> On 03/06/2017 03:02 PM, John Nash wrote: >>>>> >>>>> Here is the dump >>>>> http://pastebin.com/DTEHF5Vc >>>>> >>>>> On Mon, Mar 6, 2017 at 6:20 PM, Răzvan Crainea <raz...@opensips.org> >>>>> wrote: >>>>> >>>>>> None of the "actions" you are talking about have big impact on >>>>>> private memory, but the shared one. Better do the dump and send it over >>>>>> to >>>>>> point out what is "eating" memory. >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Răzvan Crainea >>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>> >>>>>> On 03/06/2017 02:39 PM, John Nash wrote: >>>>>> >>>>>> with every call attempt it decreases. I tried some changes by >>>>>> rejecting invite before drouting call (That means after auth , >>>>>> dispatcher) >>>>>> and found memory is stable but when drouting sends Invite to external >>>>>> gateway and external gateway rejects it. Then this issue happens. >>>>>> >>>>>> Inuse transactions and active dialogs also 0. Somthing wrong >>>>>> happening in handling of failure replies. But apart from use_next_gw >>>>>> and setting some avps for CDR not much going on there. >>>>>> >>>>>> On Mon, Mar 6, 2017 at 5:54 PM, Răzvan Crainea <raz...@opensips.org> >>>>>> wrote: >>>>>> >>>>>>> Ok, so it is the first listener for the private IP that leaks. Next, >>>>>>> is the memory stabilizing in time? Or it is continously decreasing? >>>>>>> Yes, that's how you should make the dump. >>>>>>> >>>>>>> Best regards, >>>>>>> >>>>>>> Răzvan Crainea >>>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>>> >>>>>>> >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> ___ >> 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] Quest to find memory leak
only 6 or 7 calls On Tue, Mar 7, 2017 at 3:09 PM, Răzvan Crainea <raz...@opensips.org> wrote: > So I understand that after ~3K calls, that process completely runs out of > memory? > How many calls have you done before this trace: > http://pastebin.com/9Ge2NEVQ > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/07/2017 11:32 AM, John Nash wrote: > > when I check stats after a call attempt pkmem:7-free_size:: 3304280 > > In this entry with every call I see a drop of 1000 bytes around and this > never restores. > > On Tue, Mar 7, 2017 at 2:16 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> Hi, John! >> >> Again, this trace doesn't show any leak. >> Are you sure you are having a private memory leak and not a shared >> memory leak? >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/06/2017 08:09 PM, John Nash wrote: >> >> here is another trace >> http://pastebin.com/9Ge2NEVQ >> >> I see lot of alloc request but no free. >> >> On Mon, Mar 6, 2017 at 6:57 PM, John Nash <john.nash...@gmail.com> wrote: >> >>> Ok will try that. Is it possible that wrong usage of drouting may cause >>> this to happen instead of actual leak?... What are the things private >>> memory is used for? >>> >>> On Mon, Mar 6, 2017 at 6:48 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> Hi, John! >>>> >>>> From the dump you sent, I don't see any leaks. Perhaps some of those >>>> fragments increase over time. Can you make a memory dump after the server >>>> runs some time, like after it gets 100 messages? >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 03/06/2017 03:02 PM, John Nash wrote: >>>> >>>> Here is the dump >>>> http://pastebin.com/DTEHF5Vc >>>> >>>> On Mon, Mar 6, 2017 at 6:20 PM, Răzvan Crainea <raz...@opensips.org> >>>> wrote: >>>> >>>>> None of the "actions" you are talking about have big impact on private >>>>> memory, but the shared one. Better do the dump and send it over to point >>>>> out what is "eating" memory. >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>> >>>>> On 03/06/2017 02:39 PM, John Nash wrote: >>>>> >>>>> with every call attempt it decreases. I tried some changes by >>>>> rejecting invite before drouting call (That means after auth , dispatcher) >>>>> and found memory is stable but when drouting sends Invite to external >>>>> gateway and external gateway rejects it. Then this issue happens. >>>>> >>>>> Inuse transactions and active dialogs also 0. Somthing wrong happening >>>>> in handling of failure replies. But apart from use_next_gw and >>>>> setting some avps for CDR not much going on there. >>>>> >>>>> On Mon, Mar 6, 2017 at 5:54 PM, Răzvan Crainea <raz...@opensips.org> >>>>> wrote: >>>>> >>>>>> Ok, so it is the first listener for the private IP that leaks. Next, >>>>>> is the memory stabilizing in time? Or it is continously decreasing? >>>>>> Yes, that's how you should make the dump. >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Răzvan Crainea >>>>>> OpenSIPS Solutionswww.opensips-solutions.com >>>>>> >>>>>> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > 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] Quest to find memory leak
when I check stats after a call attempt pkmem:7-free_size:: 3304280 In this entry with every call I see a drop of 1000 bytes around and this never restores. On Tue, Mar 7, 2017 at 2:16 PM, Răzvan Crainea <raz...@opensips.org> wrote: > Hi, John! > > Again, this trace doesn't show any leak. > Are you sure you are having a private memory leak and not a shared > memory leak? > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/06/2017 08:09 PM, John Nash wrote: > > here is another trace > http://pastebin.com/9Ge2NEVQ > > I see lot of alloc request but no free. > > On Mon, Mar 6, 2017 at 6:57 PM, John Nash <john.nash...@gmail.com> wrote: > >> Ok will try that. Is it possible that wrong usage of drouting may cause >> this to happen instead of actual leak?... What are the things private >> memory is used for? >> >> On Mon, Mar 6, 2017 at 6:48 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> Hi, John! >>> >>> From the dump you sent, I don't see any leaks. Perhaps some of those >>> fragments increase over time. Can you make a memory dump after the server >>> runs some time, like after it gets 100 messages? >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/06/2017 03:02 PM, John Nash wrote: >>> >>> Here is the dump >>> http://pastebin.com/DTEHF5Vc >>> >>> On Mon, Mar 6, 2017 at 6:20 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> None of the "actions" you are talking about have big impact on private >>>> memory, but the shared one. Better do the dump and send it over to point >>>> out what is "eating" memory. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 03/06/2017 02:39 PM, John Nash wrote: >>>> >>>> with every call attempt it decreases. I tried some changes by rejecting >>>> invite before drouting call (That means after auth , dispatcher) and found >>>> memory is stable but when drouting sends Invite to external gateway and >>>> external gateway rejects it. Then this issue happens. >>>> >>>> Inuse transactions and active dialogs also 0. Somthing wrong happening >>>> in handling of failure replies. But apart from use_next_gw and setting >>>> some avps for CDR not much going on there. >>>> >>>> On Mon, Mar 6, 2017 at 5:54 PM, Răzvan Crainea <raz...@opensips.org> >>>> wrote: >>>> >>>>> Ok, so it is the first listener for the private IP that leaks. Next, >>>>> is the memory stabilizing in time? Or it is continously decreasing? >>>>> Yes, that's how you should make the dump. >>>>> >>>>> Best regards, >>>>> >>>>> Răzvan Crainea >>>>> OpenSIPS Solutionswww.opensips-solutions.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] Quest to find memory leak
here is another trace http://pastebin.com/9Ge2NEVQ I see lot of alloc request but no free. On Mon, Mar 6, 2017 at 6:57 PM, John Nash <john.nash...@gmail.com> wrote: > Ok will try that. Is it possible that wrong usage of drouting may cause > this to happen instead of actual leak?... What are the things private > memory is used for? > > On Mon, Mar 6, 2017 at 6:48 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> Hi, John! >> >> From the dump you sent, I don't see any leaks. Perhaps some of those >> fragments increase over time. Can you make a memory dump after the server >> runs some time, like after it gets 100 messages? >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/06/2017 03:02 PM, John Nash wrote: >> >> Here is the dump >> http://pastebin.com/DTEHF5Vc >> >> On Mon, Mar 6, 2017 at 6:20 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> None of the "actions" you are talking about have big impact on private >>> memory, but the shared one. Better do the dump and send it over to point >>> out what is "eating" memory. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/06/2017 02:39 PM, John Nash wrote: >>> >>> with every call attempt it decreases. I tried some changes by rejecting >>> invite before drouting call (That means after auth , dispatcher) and found >>> memory is stable but when drouting sends Invite to external gateway and >>> external gateway rejects it. Then this issue happens. >>> >>> Inuse transactions and active dialogs also 0. Somthing wrong happening >>> in handling of failure replies. But apart from use_next_gw and setting >>> some avps for CDR not much going on there. >>> >>> On Mon, Mar 6, 2017 at 5:54 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> Ok, so it is the first listener for the private IP that leaks. Next, is >>>> the memory stabilizing in time? Or it is continously decreasing? >>>> Yes, that's how you should make the dump. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>>> >>>> On 03/06/2017 10:57 AM, John Nash wrote: >>>> >>>> Dear Razvan, >>>> >>>> Below is the info on my processes >>>> Process:: ID=0 PID=17351 Type=attendant >>>> Process:: ID=1 PID=17352 Type=MI FIFO >>>> Process:: ID=2 PID=17353 Type=MI Datagram >>>> Process:: ID=3 PID=17354 Type=time_keeper >>>> Process:: ID=4 PID=17355 Type=timer >>>> Process:: ID=5 PID=17356 Type=SIP receiver udp:1.1.1.1:9094 >>>> Process:: ID=6 PID=17357 Type=SIP receiver udp:1.1.1.1:5060 >>>> Process:: ID=7 PID=17358 Type=SIP receiver udp:192.168.45.5:5064 >>>> Process:: ID=8 PID=17359 Type=Timer handler >>>> >>>> 1.1.1.1 is public IP (I changed). The decrease in memory I see is for >>>> Process:: ID=7 PID=17358 mainly. My call flow is as following >>>> >>>> - New Invite hits the opensips on 1.1.1.1:9094 >>>> - Apart from message validity checks I query DB to check if its a valid >>>> user (Using local cache also there) >>>> - Create dialog, Topology_hiding functions are called along with some >>>> avp population >>>> - Using dispatcher ds_select_domain Call sent to udp:192.168.45.2:7060 >>>> (using force socket). This 192.168.45.2:7060 is actually freeswitch >>>> - Call again comes back to opensips on udp:192.168.45.5:5064 >>>> - New dialog is created and topology_hiding is called >>>> - Drouting function do_routing is called which tries one gateway and >>>> fails >>>> >>>> >>>> Dump i need to create with memlog=4 memdump=1 right? >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On Mon, Mar 6, 2017 at 2:05 PM, Răzvan Crainea <raz...@opensips.org> >>>> wrote: >>>> >>>>> Hi, John! >>>>> >>>>> Transactions are stored in shared memory, not in the private one. So >>>>> the possible leak you are facing its not related to transactions. >>>>> During runtime,
Re: [OpenSIPS-Users] Quest to find memory leak
Ok will try that. Is it possible that wrong usage of drouting may cause this to happen instead of actual leak?... What are the things private memory is used for? On Mon, Mar 6, 2017 at 6:48 PM, Răzvan Crainea <raz...@opensips.org> wrote: > Hi, John! > > From the dump you sent, I don't see any leaks. Perhaps some of those > fragments increase over time. Can you make a memory dump after the server > runs some time, like after it gets 100 messages? > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/06/2017 03:02 PM, John Nash wrote: > > Here is the dump > http://pastebin.com/DTEHF5Vc > > On Mon, Mar 6, 2017 at 6:20 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> None of the "actions" you are talking about have big impact on private >> memory, but the shared one. Better do the dump and send it over to point >> out what is "eating" memory. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/06/2017 02:39 PM, John Nash wrote: >> >> with every call attempt it decreases. I tried some changes by rejecting >> invite before drouting call (That means after auth , dispatcher) and found >> memory is stable but when drouting sends Invite to external gateway and >> external gateway rejects it. Then this issue happens. >> >> Inuse transactions and active dialogs also 0. Somthing wrong happening in >> handling of failure replies. But apart from use_next_gw and setting some >> avps for CDR not much going on there. >> >> On Mon, Mar 6, 2017 at 5:54 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> Ok, so it is the first listener for the private IP that leaks. Next, is >>> the memory stabilizing in time? Or it is continously decreasing? >>> Yes, that's how you should make the dump. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/06/2017 10:57 AM, John Nash wrote: >>> >>> Dear Razvan, >>> >>> Below is the info on my processes >>> Process:: ID=0 PID=17351 Type=attendant >>> Process:: ID=1 PID=17352 Type=MI FIFO >>> Process:: ID=2 PID=17353 Type=MI Datagram >>> Process:: ID=3 PID=17354 Type=time_keeper >>> Process:: ID=4 PID=17355 Type=timer >>> Process:: ID=5 PID=17356 Type=SIP receiver udp:1.1.1.1:9094 >>> Process:: ID=6 PID=17357 Type=SIP receiver udp:1.1.1.1:5060 >>> Process:: ID=7 PID=17358 Type=SIP receiver udp:192.168.45.5:5064 >>> Process:: ID=8 PID=17359 Type=Timer handler >>> >>> 1.1.1.1 is public IP (I changed). The decrease in memory I see is for >>> Process:: ID=7 PID=17358 mainly. My call flow is as following >>> >>> - New Invite hits the opensips on 1.1.1.1:9094 >>> - Apart from message validity checks I query DB to check if its a valid >>> user (Using local cache also there) >>> - Create dialog, Topology_hiding functions are called along with some >>> avp population >>> - Using dispatcher ds_select_domain Call sent to udp:192.168.45.2:7060 >>> (using force socket). This 192.168.45.2:7060 is actually freeswitch >>> - Call again comes back to opensips on udp:192.168.45.5:5064 >>> - New dialog is created and topology_hiding is called >>> - Drouting function do_routing is called which tries one gateway and >>> fails >>> >>> >>> Dump i need to create with memlog=4 memdump=1 right? >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mon, Mar 6, 2017 at 2:05 PM, Răzvan Crainea <raz...@opensips.org> >>> wrote: >>> >>>> Hi, John! >>>> >>>> Transactions are stored in shared memory, not in the private one. So >>>> the possible leak you are facing its not related to transactions. >>>> During runtime, OpenSIPS might resize some internal structures, which >>>> may lead to increase memory usage. However, after a while, these >>>> allocations should stabilize . >>>> Can you post the output of the kill -SIGUSR1 on pastebin so we can take >>>> a look? Also, what type of process is the one you are seeing the leak into? >>>> You can find out using the 'opensipsctl ps' command. >>>> >>>> Best regards, >>>> >>>> Răzvan Crainea >>>> OpenSIPS Solutionswww.opensips-solutions.com >>
Re: [OpenSIPS-Users] Quest to find memory leak
Here is the dump http://pastebin.com/DTEHF5Vc On Mon, Mar 6, 2017 at 6:20 PM, Răzvan Crainea <raz...@opensips.org> wrote: > None of the "actions" you are talking about have big impact on private > memory, but the shared one. Better do the dump and send it over to point > out what is "eating" memory. > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/06/2017 02:39 PM, John Nash wrote: > > with every call attempt it decreases. I tried some changes by rejecting > invite before drouting call (That means after auth , dispatcher) and found > memory is stable but when drouting sends Invite to external gateway and > external gateway rejects it. Then this issue happens. > > Inuse transactions and active dialogs also 0. Somthing wrong happening in > handling of failure replies. But apart from use_next_gw and setting some > avps for CDR not much going on there. > > On Mon, Mar 6, 2017 at 5:54 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> Ok, so it is the first listener for the private IP that leaks. Next, is >> the memory stabilizing in time? Or it is continously decreasing? >> Yes, that's how you should make the dump. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/06/2017 10:57 AM, John Nash wrote: >> >> Dear Razvan, >> >> Below is the info on my processes >> Process:: ID=0 PID=17351 Type=attendant >> Process:: ID=1 PID=17352 Type=MI FIFO >> Process:: ID=2 PID=17353 Type=MI Datagram >> Process:: ID=3 PID=17354 Type=time_keeper >> Process:: ID=4 PID=17355 Type=timer >> Process:: ID=5 PID=17356 Type=SIP receiver udp:1.1.1.1:9094 >> Process:: ID=6 PID=17357 Type=SIP receiver udp:1.1.1.1:5060 >> Process:: ID=7 PID=17358 Type=SIP receiver udp:192.168.45.5:5064 >> Process:: ID=8 PID=17359 Type=Timer handler >> >> 1.1.1.1 is public IP (I changed). The decrease in memory I see is for >> Process:: ID=7 PID=17358 mainly. My call flow is as following >> >> - New Invite hits the opensips on 1.1.1.1:9094 >> - Apart from message validity checks I query DB to check if its a valid >> user (Using local cache also there) >> - Create dialog, Topology_hiding functions are called along with some avp >> population >> - Using dispatcher ds_select_domain Call sent to udp:192.168.45.2:7060 >> (using force socket). This 192.168.45.2:7060 is actually freeswitch >> - Call again comes back to opensips on udp:192.168.45.5:5064 >> - New dialog is created and topology_hiding is called >> - Drouting function do_routing is called which tries one gateway and fails >> >> >> Dump i need to create with memlog=4 memdump=1 right? >> >> >> >> >> >> >> >> >> >> On Mon, Mar 6, 2017 at 2:05 PM, Răzvan Crainea <raz...@opensips.org> >> wrote: >> >>> Hi, John! >>> >>> Transactions are stored in shared memory, not in the private one. So the >>> possible leak you are facing its not related to transactions. >>> During runtime, OpenSIPS might resize some internal structures, which >>> may lead to increase memory usage. However, after a while, these >>> allocations should stabilize . >>> Can you post the output of the kill -SIGUSR1 on pastebin so we can take >>> a look? Also, what type of process is the one you are seeing the leak into? >>> You can find out using the 'opensipsctl ps' command. >>> >>> Best regards, >>> >>> Răzvan Crainea >>> OpenSIPS Solutionswww.opensips-solutions.com >>> >>> On 03/06/2017 09:55 AM, John Nash wrote: >>> >>> I am using OpenSIPS (2.1.5 (x86_64/linux)) in production. I observed >>> private memory is decreasing constantly for one process mainly and >>> ultimately leading to memory errors and crash. >>> >>> To debug this issue I prepared a test server and compiled opensips as >>> per https://www.opensips.org/Documentation/TroubleShooting-OutOfMem >>> >>> I made only one single call (which was rejected by opensips as it was >>> not authorized user) and I saw private free memory decreased. I was hoping >>> since transaction is done ideally it should release memory and should show >>> me same memory as startup but it did not. I verified this with many call >>> attempts and i see free memory is always decreasing slowly. >>> >>> I used kill -SIGUSR1 to create memory dump. But i am unable >>> to make sense of it. It shows l
Re: [OpenSIPS-Users] Quest to find memory leak
with every call attempt it decreases. I tried some changes by rejecting invite before drouting call (That means after auth , dispatcher) and found memory is stable but when drouting sends Invite to external gateway and external gateway rejects it. Then this issue happens. Inuse transactions and active dialogs also 0. Somthing wrong happening in handling of failure replies. But apart from use_next_gw and setting some avps for CDR not much going on there. On Mon, Mar 6, 2017 at 5:54 PM, Răzvan Crainea <raz...@opensips.org> wrote: > Ok, so it is the first listener for the private IP that leaks. Next, is > the memory stabilizing in time? Or it is continously decreasing? > Yes, that's how you should make the dump. > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/06/2017 10:57 AM, John Nash wrote: > > Dear Razvan, > > Below is the info on my processes > Process:: ID=0 PID=17351 Type=attendant > Process:: ID=1 PID=17352 Type=MI FIFO > Process:: ID=2 PID=17353 Type=MI Datagram > Process:: ID=3 PID=17354 Type=time_keeper > Process:: ID=4 PID=17355 Type=timer > Process:: ID=5 PID=17356 Type=SIP receiver udp:1.1.1.1:9094 > Process:: ID=6 PID=17357 Type=SIP receiver udp:1.1.1.1:5060 > Process:: ID=7 PID=17358 Type=SIP receiver udp:192.168.45.5:5064 > Process:: ID=8 PID=17359 Type=Timer handler > > 1.1.1.1 is public IP (I changed). The decrease in memory I see is for > Process:: ID=7 PID=17358 mainly. My call flow is as following > > - New Invite hits the opensips on 1.1.1.1:9094 > - Apart from message validity checks I query DB to check if its a valid > user (Using local cache also there) > - Create dialog, Topology_hiding functions are called along with some avp > population > - Using dispatcher ds_select_domain Call sent to udp:192.168.45.2:7060 > (using force socket). This 192.168.45.2:7060 is actually freeswitch > - Call again comes back to opensips on udp:192.168.45.5:5064 > - New dialog is created and topology_hiding is called > - Drouting function do_routing is called which tries one gateway and fails > > > Dump i need to create with memlog=4 memdump=1 right? > > > > > > > > > > On Mon, Mar 6, 2017 at 2:05 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> Hi, John! >> >> Transactions are stored in shared memory, not in the private one. So the >> possible leak you are facing its not related to transactions. >> During runtime, OpenSIPS might resize some internal structures, which may >> lead to increase memory usage. However, after a while, these allocations >> should stabilize . >> Can you post the output of the kill -SIGUSR1 on pastebin so we can take a >> look? Also, what type of process is the one you are seeing the leak into? >> You can find out using the 'opensipsctl ps' command. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutionswww.opensips-solutions.com >> >> On 03/06/2017 09:55 AM, John Nash wrote: >> >> I am using OpenSIPS (2.1.5 (x86_64/linux)) in production. I observed >> private memory is decreasing constantly for one process mainly and >> ultimately leading to memory errors and crash. >> >> To debug this issue I prepared a test server and compiled opensips as per >> https://www.opensips.org/Documentation/TroubleShooting-OutOfMem >> >> I made only one single call (which was rejected by opensips as it was not >> authorized user) and I saw private free memory decreased. I was hoping >> since transaction is done ideally it should release memory and should show >> me same memory as startup but it did not. I verified this with many call >> attempts and i see free memory is always decreasing slowly. >> >> I used kill -SIGUSR1 to create memory dump. But i am unable >> to make sense of it. It shows log like ... >> >> r 6 07:29:19 Server3021 opensips[13276]: Memory status (pkg): >> Mar 6 07:29:19 Server3021 opensips[13276]: qm_status (0x7f5b8ebba010): >> Mar 6 07:29:19 Server3021 opensips[13276]: heap size= 4194304 >> Mar 6 07:29:19 Server3021 opensips[13276]: used= 346768, >> used+overhead=848792, free=3345512 >> Mar 6 07:29:19 Server3021 opensips[13276]: max used (+overhead)= 931920 >> Mar 6 07:29:19 Server3021 opensips[13276]: dumping all alloc'ed. >> fragments: >> Mar 6 07:29:19 Server3021 opensips[13276]: 0. N >> address=0x7f5b8ebef528 frag=0x7f5b8ebef4f8 size=40 used=1 >> Mar 6 07:29:19 Server3021 opensips[13276]: alloc'd from >> script_cb.c: add_callback(60) >> Mar 6 07:29:19 Server3021 opensips[13276]: start >> check=f0f0f0f0f0f0
Re: [OpenSIPS-Users] Quest to find memory leak
Dear Razvan, Below is the info on my processes Process:: ID=0 PID=17351 Type=attendant Process:: ID=1 PID=17352 Type=MI FIFO Process:: ID=2 PID=17353 Type=MI Datagram Process:: ID=3 PID=17354 Type=time_keeper Process:: ID=4 PID=17355 Type=timer Process:: ID=5 PID=17356 Type=SIP receiver udp:1.1.1.1:9094 Process:: ID=6 PID=17357 Type=SIP receiver udp:1.1.1.1:5060 Process:: ID=7 PID=17358 Type=SIP receiver udp:192.168.45.5:5064 Process:: ID=8 PID=17359 Type=Timer handler 1.1.1.1 is public IP (I changed). The decrease in memory I see is for Process:: ID=7 PID=17358 mainly. My call flow is as following - New Invite hits the opensips on 1.1.1.1:9094 - Apart from message validity checks I query DB to check if its a valid user (Using local cache also there) - Create dialog, Topology_hiding functions are called along with some avp population - Using dispatcher ds_select_domain Call sent to udp:192.168.45.2:7060 (using force socket). This 192.168.45.2:7060 is actually freeswitch - Call again comes back to opensips on udp:192.168.45.5:5064 - New dialog is created and topology_hiding is called - Drouting function do_routing is called which tries one gateway and fails Dump i need to create with memlog=4 memdump=1 right? On Mon, Mar 6, 2017 at 2:05 PM, Răzvan Crainea <raz...@opensips.org> wrote: > Hi, John! > > Transactions are stored in shared memory, not in the private one. So the > possible leak you are facing its not related to transactions. > During runtime, OpenSIPS might resize some internal structures, which may > lead to increase memory usage. However, after a while, these allocations > should stabilize . > Can you post the output of the kill -SIGUSR1 on pastebin so we can take a > look? Also, what type of process is the one you are seeing the leak into? > You can find out using the 'opensipsctl ps' command. > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 03/06/2017 09:55 AM, John Nash wrote: > > I am using OpenSIPS (2.1.5 (x86_64/linux)) in production. I observed > private memory is decreasing constantly for one process mainly and > ultimately leading to memory errors and crash. > > To debug this issue I prepared a test server and compiled opensips as per > https://www.opensips.org/Documentation/TroubleShooting-OutOfMem > > I made only one single call (which was rejected by opensips as it was not > authorized user) and I saw private free memory decreased. I was hoping > since transaction is done ideally it should release memory and should show > me same memory as startup but it did not. I verified this with many call > attempts and i see free memory is always decreasing slowly. > > I used kill -SIGUSR1 to create memory dump. But i am unable > to make sense of it. It shows log like ... > > r 6 07:29:19 Server3021 opensips[13276]: Memory status (pkg): > Mar 6 07:29:19 Server3021 opensips[13276]: qm_status (0x7f5b8ebba010): > Mar 6 07:29:19 Server3021 opensips[13276]: heap size= 4194304 > Mar 6 07:29:19 Server3021 opensips[13276]: used= 346768, > used+overhead=848792, free=3345512 > Mar 6 07:29:19 Server3021 opensips[13276]: max used (+overhead)= 931920 > Mar 6 07:29:19 Server3021 opensips[13276]: dumping all alloc'ed. > fragments: > Mar 6 07:29:19 Server3021 opensips[13276]: 0. N > address=0x7f5b8ebef528 frag=0x7f5b8ebef4f8 size=40 used=1 > Mar 6 07:29:19 Server3021 opensips[13276]: alloc'd from > script_cb.c: add_callback(60) > Mar 6 07:29:19 Server3021 opensips[13276]: start > check=f0f0f0f0f0f0f0f0, end check= c0c0c0c0c0c0c0c0, abcdefedabcdefed > Mar 6 07:29:19 Server3021 opensips[13276]: 1. N > address=0x7f5b8ebef5b0 > > I pasted only few lines in this mail. What should be my next step?...How > can i really trace what is wrong in my script or any other memory leak? > > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > 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] Quest to find memory leak
I am using OpenSIPS (2.1.5 (x86_64/linux)) in production. I observed private memory is decreasing constantly for one process mainly and ultimately leading to memory errors and crash. To debug this issue I prepared a test server and compiled opensips as per https://www.opensips.org/Documentation/TroubleShooting-OutOfMem I made only one single call (which was rejected by opensips as it was not authorized user) and I saw private free memory decreased. I was hoping since transaction is done ideally it should release memory and should show me same memory as startup but it did not. I verified this with many call attempts and i see free memory is always decreasing slowly. I used kill -SIGUSR1 to create memory dump. But i am unable to make sense of it. It shows log like ... r 6 07:29:19 Server3021 opensips[13276]: Memory status (pkg): Mar 6 07:29:19 Server3021 opensips[13276]: qm_status (0x7f5b8ebba010): Mar 6 07:29:19 Server3021 opensips[13276]: heap size= 4194304 Mar 6 07:29:19 Server3021 opensips[13276]: used= 346768, used+overhead=848792, free=3345512 Mar 6 07:29:19 Server3021 opensips[13276]: max used (+overhead)= 931920 Mar 6 07:29:19 Server3021 opensips[13276]: dumping all alloc'ed. fragments: Mar 6 07:29:19 Server3021 opensips[13276]: 0. N address=0x7f5b8ebef528 frag=0x7f5b8ebef4f8 size=40 used=1 Mar 6 07:29:19 Server3021 opensips[13276]: alloc'd from script_cb.c: add_callback(60) Mar 6 07:29:19 Server3021 opensips[13276]: start check=f0f0f0f0f0f0f0f0, end check= c0c0c0c0c0c0c0c0, abcdefedabcdefed Mar 6 07:29:19 Server3021 opensips[13276]: 1. N address=0x7f5b8ebef5b0 I pasted only few lines in this mail. What should be my next step?...How can i really trace what is wrong in my script or any other memory leak? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dispatcher module gateway status
I am using opensips version 2.1 and using dispatcher module. In fact i have been using it for a while without any problem. But today suddenly all my gateways started showing status as "probing". I took wireshark trace and can see that opensips is pinging gateways (freeswitch) and getting reply as 200 OK but gateway status still shows probing. Any ideas where to check? (already gone through log files of both opensips and freeswitch and nothing related to dispatcher shows) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
Yes fingerprints are different in Invite and session progress. On Fri, Jun 24, 2016 at 3:00 AM, sevpal <sev...@aol.com> wrote: > Take a look at the “fingerprint:” line. > > *From:* John Nash <john.nash...@gmail.com> > *Sent:* Thursday, June 23, 2016 3:42 PM > *To:* OpenSIPS users mailling list <users@lists.opensips.org> > *Subject:* Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc > > Actually the issue is i hear no audio on either side and just after > session progress (I guess when media starts coming from remote media > server) i see error "SRTP output wanted, but no crypto suite was > negotiated" > > I had also checked media logs i could see RTP packets being sent from > freeswitch to RTPengine IP but there was no packet at all just after that. > Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should > send that packet to browser using wss? > > On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme <e...@uphreak.com> wrote: > >> So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and >> Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the >> invite with an answer in the 183, and in the 200. What is the failure you >> are seeing, and where is it happening (in freeswitch? in the browser?) >> >> The only thing that looks bad is that you are retransmitting the ACK >> which FS either ... doesnt like, or is never getting, because it keeps >> retransmitting the 200, which is why you get a 481 when you send BYE. >> >> -Eric >> >> >> On 06/23/2016 01:24 PM, John Nash wrote: >> >> OK here is the log >> https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 >> >> Sorry took me a while to convert wireshark trace to text file. >> >> My freeswitch is running on private IP (127.0.0.1) and opensips I run on >> both public and private so that for outside world opensips is the only >> public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and >> back. >> >> >> >> >> >> >> On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme <e...@uphreak.com> wrote: >> >>> No - it's annoying to look at a trace that's had information removed and >>> try and piece together whats happening. Your paranoid side is wrong, sorry. >>> >>> -Eric >>> >>> >>> On 06/23/2016 01:06 PM, Patrick Wakano wrote: >>> >>> my paranoic side would recommend to hide/change private informations, >>> specially any authentication line that might appear... this is certainly a >>> sort of social engineering threat we should worry... >>> better be safe than sorry >>> >>> >>> On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme <e...@uphreak.com> wrote: >>> >>>> I mean you can use a private gist, but you will be publishing the link >>>> in a public email list. In general I personally dont believe revealing ip >>>> addresses etc. is any problem - to put my money where my mouth is here is a >>>> gist link to an unaltered SIP trace on my server :) >>>> >>>> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >>>> >>>> -Eric >>>> >>>> >>>> On 06/23/2016 12:23 PM, John Nash wrote: >>>> >>>> Ok i am ready with logs. About gist may I use private option as traces >>>> have our IPs, user >>>> >>>> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme <e...@uphreak.com> wrote: >>>> >>>>> Hey John, >>>>> >>>>> Please paste a full UNALTERED sip trace into a gist (gist.github.com) >>>>> from the proxy servers perspective and provide a link so that we can see >>>>> what comes in, and what goes out from both sides. >>>>> >>>>> EG: ngrep -qtd any -W byline port 5060 >>>>> >>>>> This will show us the traffic that is leaving the proxy destined for >>>>> the Freeswitch box, and what the freeswitch box sends back. >>>>> >>>>> Also - you can look in your browsers console log and provide the SIP >>>>> trace from there in a seperate gist, so that we can see what opensips >>>>> sends >>>>> back up to your browser. >>>>> >>>>> -Eric >>>>> >>>>> >>>>> Am I using correct sip.js example? I copied it to my server and >>>>> accessing it using https: (used letsencrypt)
Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
Sorry sevpal somehow your message went to spam. I am not sure I get what you are trying to say as I was under the impression rtpengine is supposed to bridge protocols. Better I explain my test setup properly to you .. 1- On linux server I installed certificates from letsencrypt for a domain (). 2- I have opensips (wss listner is there as well as udp), Rtpengine and freeswitch (udp only and it terminate calls to SIP network) 3- On web server I copied sipml5 code which I access on chrome browser using https://:443. In sipml5 I give wss url of the opensips wss listener (wss://:4431 along with SIP credentials 4- My call flow is Chrome(sipml5) ==wss==>Opensips===udp==>Freeswitch. Before sending Invite to freeswitch Rtpengine call is made as per http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2. Same being done when session progress or 200 OK comes from freeswitch. Now in this setup how can I make sure same crypto is used? On Fri, Jun 24, 2016 at 2:50 AM, sevpal <sev...@aol.com> wrote: > Hi, the rtpengine cannot negotiate SRTP between the two points, both must > support the same cryptography and protocol. eg; SRTP to SRTP , DTLS/SRTP to > DTLS/SRTP cipher 128 to 128 and 256 to 256. > > You can print the request body ($rb) on the INVITE with “application/sdp” > and visually compare the exchange, do this on offer and answer. > > *From:* John Nash <john.nash...@gmail.com> > *Sent:* Thursday, June 23, 2016 3:42 PM > *To:* OpenSIPS users mailling list <users@lists.opensips.org> > *Subject:* Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc > > Actually the issue is i hear no audio on either side and just after > session progress (I guess when media starts coming from remote media > server) i see error "SRTP output wanted, but no crypto suite was > negotiated" > > I had also checked media logs i could see RTP packets being sent from > freeswitch to RTPengine IP but there was no packet at all just after that. > Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should > send that packet to browser using wss? > > On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme <e...@uphreak.com> wrote: > >> So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and >> Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the >> invite with an answer in the 183, and in the 200. What is the failure you >> are seeing, and where is it happening (in freeswitch? in the browser?) >> >> The only thing that looks bad is that you are retransmitting the ACK >> which FS either ... doesnt like, or is never getting, because it keeps >> retransmitting the 200, which is why you get a 481 when you send BYE. >> >> -Eric >> >> >> On 06/23/2016 01:24 PM, John Nash wrote: >> >> OK here is the log >> https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 >> >> Sorry took me a while to convert wireshark trace to text file. >> >> My freeswitch is running on private IP (127.0.0.1) and opensips I run on >> both public and private so that for outside world opensips is the only >> public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and >> back. >> >> >> >> >> >> >> On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme <e...@uphreak.com> wrote: >> >>> No - it's annoying to look at a trace that's had information removed and >>> try and piece together whats happening. Your paranoid side is wrong, sorry. >>> >>> -Eric >>> >>> >>> On 06/23/2016 01:06 PM, Patrick Wakano wrote: >>> >>> my paranoic side would recommend to hide/change private informations, >>> specially any authentication line that might appear... this is certainly a >>> sort of social engineering threat we should worry... >>> better be safe than sorry >>> >>> >>> On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme <e...@uphreak.com> wrote: >>> >>>> I mean you can use a private gist, but you will be publishing the link >>>> in a public email list. In general I personally dont believe revealing ip >>>> addresses etc. is any problem - to put my money where my mouth is here is a >>>> gist link to an unaltered SIP trace on my server :) >>>> >>>> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >>>> >>>> -Eric >>>> >>>> >>>> On 06/23/2016 12:23 PM, John Nash wrote: >>>> >>>> Ok i am ready with logs. About gist may I use private option as traces >>>> have our IPs, user >>>> &
Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
Actually the issue is i hear no audio on either side and just after session progress (I guess when media starts coming from remote media server) i see error "SRTP output wanted, but no crypto suite was negotiated" I had also checked media logs i could see RTP packets being sent from freeswitch to RTPengine IP but there was no packet at all just after that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine should send that packet to browser using wss? On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme <e...@uphreak.com> wrote: > So - i dont see a problem here - Chrome is getting UDP/TLS/RTP/SAVPF and > Freeswitch is getting RTP/AVP. Freeswitch responded to the offer in the > invite with an answer in the 183, and in the 200. What is the failure you > are seeing, and where is it happening (in freeswitch? in the browser?) > > The only thing that looks bad is that you are retransmitting the ACK which > FS either ... doesnt like, or is never getting, because it keeps > retransmitting the 200, which is why you get a 481 when you send BYE. > > -Eric > > > On 06/23/2016 01:24 PM, John Nash wrote: > > OK here is the log > https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 > > Sorry took me a while to convert wireshark trace to text file. > > My freeswitch is running on private IP (127.0.0.1) and opensips I run on > both public and private so that for outside world opensips is the only > public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and > back. > > > > > > > On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme <e...@uphreak.com> wrote: > >> No - it's annoying to look at a trace that's had information removed and >> try and piece together whats happening. Your paranoid side is wrong, sorry. >> >> -Eric >> >> >> On 06/23/2016 01:06 PM, Patrick Wakano wrote: >> >> my paranoic side would recommend to hide/change private informations, >> specially any authentication line that might appear... this is certainly a >> sort of social engineering threat we should worry... >> better be safe than sorry >> >> >> On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme < <e...@uphreak.com> >> e...@uphreak.com> wrote: >> >>> I mean you can use a private gist, but you will be publishing the link >>> in a public email list. In general I personally dont believe revealing ip >>> addresses etc. is any problem - to put my money where my mouth is here is a >>> gist link to an unaltered SIP trace on my server :) >>> >>> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >>> >>> -Eric >>> >>> >>> On 06/23/2016 12:23 PM, John Nash wrote: >>> >>> Ok i am ready with logs. About gist may I use private option as traces >>> have our IPs, user >>> >>> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme < <e...@uphreak.com> >>> e...@uphreak.com> wrote: >>> >>>> Hey John, >>>> >>>> Please paste a full UNALTERED sip trace into a gist (gist.github.com) >>>> from the proxy servers perspective and provide a link so that we can see >>>> what comes in, and what goes out from both sides. >>>> >>>> EG: ngrep -qtd any -W byline port 5060 >>>> >>>> This will show us the traffic that is leaving the proxy destined for >>>> the Freeswitch box, and what the freeswitch box sends back. >>>> >>>> Also - you can look in your browsers console log and provide the SIP >>>> trace from there in a seperate gist, so that we can see what opensips sends >>>> back up to your browser. >>>> >>>> -Eric >>>> >>>> >>>> Am I using correct sip.js example? I copied it to my server and >>>> accessing it using https: (used letsencrypt) >>>> >>>> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme < <e...@uphreak.com> >>>> e...@uphreak.com> wrote: >>>> >>>>> 1. I would suggest using SIP.js - <https://github.com/onsip/SIP.js> >>>>> https://github.com/onsip/SIP.js it is a much more active project that >>>>> sipml5. >>>>> >>>>> 2. Im guessing that you are not properly passing flags to RTPEngine. >>>>> If you want to have DTLS-SRTP between the browser, and plain RTP/AVP >>>>> between RTPEngine and freeswitch, you need to "offer" rtp/avp to >>>>> freeswitch, and "answer" dtls-srtp back up to the browser. >>&
Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
OK here is the log https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744 Sorry took me a while to convert wireshark trace to text file. My freeswitch is running on private IP (127.0.0.1) and opensips I run on both public and private so that for outside world opensips is the only public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and back. On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme <e...@uphreak.com> wrote: > No - it's annoying to look at a trace that's had information removed and > try and piece together whats happening. Your paranoid side is wrong, sorry. > > -Eric > > > On 06/23/2016 01:06 PM, Patrick Wakano wrote: > > my paranoic side would recommend to hide/change private informations, > specially any authentication line that might appear... this is certainly a > sort of social engineering threat we should worry... > better be safe than sorry > > > On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme <e...@uphreak.com> wrote: > >> I mean you can use a private gist, but you will be publishing the link in >> a public email list. In general I personally dont believe revealing ip >> addresses etc. is any problem - to put my money where my mouth is here is a >> gist link to an unaltered SIP trace on my server :) >> >> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52 >> >> -Eric >> >> >> On 06/23/2016 12:23 PM, John Nash wrote: >> >> Ok i am ready with logs. About gist may I use private option as traces >> have our IPs, user >> >> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme < <e...@uphreak.com> >> e...@uphreak.com> wrote: >> >>> Hey John, >>> >>> Please paste a full UNALTERED sip trace into a gist (gist.github.com) >>> from the proxy servers perspective and provide a link so that we can see >>> what comes in, and what goes out from both sides. >>> >>> EG: ngrep -qtd any -W byline port 5060 >>> >>> This will show us the traffic that is leaving the proxy destined for the >>> Freeswitch box, and what the freeswitch box sends back. >>> >>> Also - you can look in your browsers console log and provide the SIP >>> trace from there in a seperate gist, so that we can see what opensips sends >>> back up to your browser. >>> >>> -Eric >>> >>> >>> Am I using correct sip.js example? I copied it to my server and >>> accessing it using https: (used letsencrypt) >>> >>> On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme < <e...@uphreak.com> >>> e...@uphreak.com> wrote: >>> >>>> 1. I would suggest using SIP.js - <https://github.com/onsip/SIP.js> >>>> https://github.com/onsip/SIP.js it is a much more active project that >>>> sipml5. >>>> >>>> 2. Im guessing that you are not properly passing flags to RTPEngine. >>>> If you want to have DTLS-SRTP between the browser, and plain RTP/AVP >>>> between RTPEngine and freeswitch, you need to "offer" rtp/avp to >>>> freeswitch, and "answer" dtls-srtp back up to the browser. >>>> >>>> the offer to freeswitch would be: >>>> >>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection >>>> replace-origin ICE=remove"; >>>> >>>> >>>> and the answer back up to the browswer would be: >>>> >>>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >>>> >>>> >>>> -Eric >>>> >>>> >>>> >>>> On 06/23/2016 08:20 AM, John Nash wrote: >>>> >>>> I am following >>>> <http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2> >>>> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and >>>> trying to test a call >>>> >>>> sipml5 --->Opensips + rtpengine > SIP end point >>>> (Freeswitch) >>>> >>>> But I do not have any audio on both sides. I see this error at >>>> rtpengine log "SRTP output wanted, but no crypto suite was negotiated" >>>> >>>> Anyone tested this scenario positive? >>>> >>>> >>>> ___ >>>> Users mailing >>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>&g
Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc
Ok i am ready with logs. About gist may I use private option as traces have our IPs, user On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme <e...@uphreak.com> wrote: > Hey John, > > Please paste a full UNALTERED sip trace into a gist (gist.github.com) > from the proxy servers perspective and provide a link so that we can see > what comes in, and what goes out from both sides. > > EG: ngrep -qtd any -W byline port 5060 > > This will show us the traffic that is leaving the proxy destined for the > Freeswitch box, and what the freeswitch box sends back. > > Also - you can look in your browsers console log and provide the SIP trace > from there in a seperate gist, so that we can see what opensips sends back > up to your browser. > > -Eric > > > Am I using correct sip.js example? I copied it to my server and accessing > it using https: (used letsencrypt) > > On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme <e...@uphreak.com> wrote: > >> 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is >> a much more active project that sipml5. >> >> 2. Im guessing that you are not properly passing flags to RTPEngine. If >> you want to have DTLS-SRTP between the browser, and plain RTP/AVP between >> RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and >> "answer" dtls-srtp back up to the browser. >> >> the offer to freeswitch would be: >> >> $var(rtpengine_flags) = "RTP/AVP replace-session-connection >> replace-origin ICE=remove"; >> >> >> and the answer back up to the browswer would be: >> >> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >> >> >> -Eric >> >> >> >> On 06/23/2016 08:20 AM, John Nash wrote: >> >> I am following >> <http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2> >> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying >> to test a call >> >> sipml5 --->Opensips + rtpengine > SIP end point >> (Freeswitch) >> >> But I do not have any audio on both sides. I see this error at rtpengine >> log "SRTP output wanted, but no crypto suite was negotiated" >> >> Anyone tested this scenario positive? >> >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > 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] Opensips + rtpengine + Sipml5 webrtc
I double checked my rtpengine offer answer calls and now using https://github.com/onsip/sipjs-examples/tree/master/demo-phone but I face same issue (no audio either side) and error "SRTP output wanted, but no crypto suite was negotiated" Rtpengine also I updated to the latest now. Am I using correct sip.js example? I copied it to my server and accessing it using https: (used letsencrypt) On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme <e...@uphreak.com> wrote: > 1. I would suggest using SIP.js - https://github.com/onsip/SIP.js it is a > much more active project that sipml5. > > 2. Im guessing that you are not properly passing flags to RTPEngine. If > you want to have DTLS-SRTP between the browser, and plain RTP/AVP between > RTPEngine and freeswitch, you need to "offer" rtp/avp to freeswitch, and > "answer" dtls-srtp back up to the browser. > > the offer to freeswitch would be: > > $var(rtpengine_flags) = "RTP/AVP replace-session-connection > replace-origin ICE=remove"; > > > and the answer back up to the browswer would be: > > $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; > > > -Eric > > > > On 06/23/2016 08:20 AM, John Nash wrote: > > I am following > http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying > to test a call > > sipml5 --->Opensips + rtpengine > SIP end point > (Freeswitch) > > But I do not have any audio on both sides. I see this error at rtpengine > log "SRTP output wanted, but no crypto suite was negotiated" > > Anyone tested this scenario positive? > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > 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 + rtpengine + Sipml5 webrtc
I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying to test a call sipml5 --->Opensips + rtpengine > SIP end point (Freeswitch) But I do not have any audio on both sides. I see this error at rtpengine log "SRTP output wanted, but no crypto suite was negotiated" Anyone tested this scenario positive? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] webrtc native client for opensips
OK but in this i will not have control over webrtc codecs. What I am looking for is ... 1- Native application can send SIP messages over wss. 2- For media instead of using chrome functions , use my custom webrtc ( https://webrtc.org/native-code/ios/) which I use as library I am not even sure at this point if I am making sense. On Wed, Jun 22, 2016 at 3:23 PM, Tito Cumpen <t...@xsvoce.com> wrote: > If you intend on running on android and iOS restcomm appears to have > native clients that support sip over websocket . I've tested the iOS app > they with OpenSIPS and baseline functionality was there . > https://github.com/RestComm/restcomm-android-sdk > On Jun 22, 2016 5:34 AM, "John Nash" <john.nash...@gmail.com> wrote: > >> My objective is to make a native webrtc application which can use SIP >> over wss for signalling and for media also I do not want to be dependent on >> chrome as in future I wish to incorporate more codecs into it. >> >> Any pointers for me? >> >> On Wed, Jun 22, 2016 at 2:55 PM, Tito Cumpen <t...@xsvoce.com> wrote: >> >>> John, >>> >>> You can utilize sipjs and jssip on account that they utilize sip over >>> websocket. Take into consideration that chrome will only allow getusermedia >>> if you are using wss and https . >>> On Jun 22, 2016 3:07 AM, "John Nash" <john.nash...@gmail.com> wrote: >>> >>>> Apart from sipml5 is there any native webrtc client also which I can >>>> explore to work with opensips? >>>> >>>> The examples I find for webrtc native seem to be using jingle protocol >>>> but in case to make it work with opensips, It has to use SIP/SDP at client >>>> end right?..Any examples? >>>> >>>> ___ >>>> 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 >>> >>> >> >> ___ >> 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 > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] webrtc native client for opensips
My objective is to make a native webrtc application which can use SIP over wss for signalling and for media also I do not want to be dependent on chrome as in future I wish to incorporate more codecs into it. Any pointers for me? On Wed, Jun 22, 2016 at 2:55 PM, Tito Cumpen <t...@xsvoce.com> wrote: > John, > > You can utilize sipjs and jssip on account that they utilize sip over > websocket. Take into consideration that chrome will only allow getusermedia > if you are using wss and https . > On Jun 22, 2016 3:07 AM, "John Nash" <john.nash...@gmail.com> wrote: > >> Apart from sipml5 is there any native webrtc client also which I can >> explore to work with opensips? >> >> The examples I find for webrtc native seem to be using jingle protocol >> but in case to make it work with opensips, It has to use SIP/SDP at client >> end right?..Any examples? >> >> ___ >> 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 > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] webrtc native client for opensips
Apart from sipml5 is there any native webrtc client also which I can explore to work with opensips? The examples I find for webrtc native seem to be using jingle protocol but in case to make it work with opensips, It has to use SIP/SDP at client end right?..Any examples? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] web sockets (wss) error
Git version (2.2) is OK. May be in tar download latest files are not there no biggi. On Tue, Jun 21, 2016 at 10:08 PM, John Nash <john.nash...@gmail.com> wrote: > I downloaded opensips 2.2 stable tar file and upgraded my existing > opensips.cfg to use wss as per document > http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 > > but when I start opensips I get error "cannot handle protocol " right > at the line where I have listener wss:127.0.0.1:443 > > Do I need to clone current git version in order to test wss? > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] web sockets (wss) error
I downloaded opensips 2.2 stable tar file and upgraded my existing opensips.cfg to use wss as per document http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 but when I start opensips I get error "cannot handle protocol " right at the line where I have listener wss:127.0.0.1:443 Do I need to clone current git version in order to test wss? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] codec_delete function issue
I tried this function to delete "NSE" codec from SDP but it doesnt seem to be working. Any consideration to make it work?...I am using rtpengine module but calling it after calling codec_delete but this codec still passes to other end. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Register with TO Tag
OK that means I should handle Register before In-dialog processing block?
I also have one more doubt function mf_process_maxfwd_header should it be
used before sipmsg_validate or after?...Currently mf_process_maxfwd_header
is being called in my script first but in some cases with malformed
packets its not even able to read max fwd header.
On Tue, Jun 7, 2016 at 4:11 PM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:
> Hi John,
>
> Assuming you do not do REGISTER relay (but you act as a registrar), you
> should handle the REGISTER requests (with or without to-tag) in the same
> way. IF they have a Route hdr , it may be because they do pre-loaded route
> (the Route points to your SIP server) to be sure the REGISTER gets to the
> registrar server.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 07.06.2016 10:56, John Nash wrote:
>
> I am dealing with In-dialog requests using
>
>
> --
> if (has_totag() && (is_domain_local("$rd") || $Ri== "127.0.0.1") &&
> is_method("INVITE|ACK|BYE|UPDATE"))
> {
> # sequential request within a dialog should
> # take the path determined by record-routing
> if (topology_hiding_match())
> -
>
> -
>
> at the top of my script. After that I process initial requests, but I see
> some REGISTER messages with TO-Tag and "Route" header and they are being
> discarded by my script because Initial request cannot have Route header.
>
> Do i also need to pass REGISTER messages also through same block?...or i
> need to call loose_route after has_to_tag check.
>
>
>
>
> ___
> Users mailing
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Register with TO Tag
I am dealing with In-dialog requests using
--
if (has_totag() && (is_domain_local("$rd") || $Ri== "127.0.0.1") &&
is_method("INVITE|ACK|BYE|UPDATE"))
{
# sequential request within a dialog should
# take the path determined by record-routing
if (topology_hiding_match())
-
-
at the top of my script. After that I process initial requests, but I see
some REGISTER messages with TO-Tag and "Route" header and they are being
discarded by my script because Initial request cannot have Route header.
Do i also need to pass REGISTER messages also through same block?...or i
need to call loose_route after has_to_tag check.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Rate limit question
Thank you for detailed reply. I will choose to implement on Opensips. I would also prefer performance over accuracy as my objective is to stop customers from flooding with too many calls so I can select RED? On Fri, Jun 3, 2016 at 12:47 PM, Răzvan Crainea <raz...@opensips.org> wrote: > Hi, John! > > It depends on the logic of your platform. If you have multiple FreeSWITCH > servers behind OpenSIPS, and you want to have a "global" limit, then you > must do it on OpenSIPS side. It is also recommended to do it there because > for example if the threashold is reached, you don't have to load FreeSWITCH > with traffic that's about to be denied. However, if you have local (per > FreeSWITCH) limits, then you can do it on the FreeSWITCH side (with the > earlier observations). > > Regarding the algorithm, it depends on what you want to limit. If you want > to do CPS, then TAILDROP or RED is more suitable. If you want to do a > limitation based on the network load, then use NETWORK. > > Also take into account that the way RED and TAILDROP algorithm work, they > are not so accurate. That's why in OpenSIPS 2.2 we added a new algorithm, > SBT[1], which is very accurate and custamizable. > > [1] > http://www.opensips.org/html/docs/modules/2.2.x/ratelimit.html#id293435 > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/02/2016 08:49 AM, John Nash wrote: > > Also how can I decide which Rate limit algorithm should I choose ? Like > RED or TAILDROP or NETWORK > > On Thu, Jun 2, 2016 at 9:37 AM, John Nash <john.nash...@gmail.com> wrote: > >> I am using opensips(2,1) + freeswitch. At opensips doing auth and >> drouting. Now i plan to test rate limit but should I be checking CPS at >> opensips or at freeswitch?...as Rate limit uses timers would it be more >> appropriate to check at freeswitch? >> >> >> > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > 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] Rate limit question
Also how can I decide which Rate limit algorithm should I choose ? Like RED or TAILDROP or NETWORK On Thu, Jun 2, 2016 at 9:37 AM, John Nash <john.nash...@gmail.com> wrote: > I am using opensips(2,1) + freeswitch. At opensips doing auth and > drouting. Now i plan to test rate limit but should I be checking CPS at > opensips or at freeswitch?...as Rate limit uses timers would it be more > appropriate to check at freeswitch? > > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Rate limit question
I am using opensips(2,1) + freeswitch. At opensips doing auth and drouting. Now i plan to test rate limit but should I be checking CPS at opensips or at freeswitch?...as Rate limit uses timers would it be more appropriate to check at freeswitch? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Multi homed setup
Thank you. I tested using manual and so far seems to be working fine. On Mon, May 30, 2016 at 7:04 PM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > Hello Nash, > > This is a limitation of how mhomed works - basically opensips asks the OS > to provide a suitable source interface to reach a certain destination. But > OS will consider all the network interface, not only the ones configured in > OpenSIPS. So, the OS may select and return an interface not used by > OpenSIPS. > > What you can do: > 1) make 1.1.1.1 default interface, to favor it in the OS selection > 2) disable mhomed and use manual selection of the interface via > force_send_socket()[1] or $fs[2] > > For each GW, dr_gateway, you can store in attributes string some info > about the interface to be used to reach that GW. > > [1] http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#toc16 > [2] http://www.opensips.org/Documentation/Script-CoreVar-2-2#toc43 > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 30.05.2016 11:33, John Nash wrote: > > On my linux box we have multiple public IP addressess > 1.1.1.1 > 2.2.2.2 > 3.3.3.3 > > I am listening on two of them as > udp:1.1.1.1:5060 > udp:2.2.2.2:5060 > > I have mhomed=1 in my config. I am also using drouting module. What I > expect is when an Invite comes to 1.1.1.1:5060 . drouting should send > outgoing invite using 1.1.1.1 as source IP but it tries to send using > 3.3.3.3 (Which is default interface) > > Is this the expected behavior of mhomed ?..should I manually try to > control this? > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://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] Multi homed setup
On my linux box we have multiple public IP addressess 1.1.1.1 2.2.2.2 3.3.3.3 I am listening on two of them as udp:1.1.1.1:5060 udp:2.2.2.2:5060 I have mhomed=1 in my config. I am also using drouting module. What I expect is when an Invite comes to 1.1.1.1:5060 . drouting should send outgoing invite using 1.1.1.1 as source IP but it tries to send using 3.3.3.3 (Which is default interface) Is this the expected behavior of mhomed ?..should I manually try to control this? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] force_send_socket arguments
wow. Cool !!! On Mon, May 30, 2016 at 1:57 PM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > Hi John, > > The function does not accept any kind of variables, but, with the same > behavior, you can use the $fs variable (instead of the function): > http://www.opensips.org/Documentation/Script-CoreVar-2-2#toc43 > > So, instead of > force_send_socket(tcp:10.10.10.10:5060); > you can do > $fs = "tcp:10.10.10.10:5060"; > > And on the right side of the assignment you can use any vars you need: > $fs = "tcp:"+$avp(my_IP)+":5060"; > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 29.05.2016 16:20, John Nash wrote: > > Is it possible to use avp or any other vraiable as argument > to force_send_socket ? > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://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] force_send_socket arguments
Is it possible to use avp or any other vraiable as argument to force_send_socket ? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Security presentation on opensips
Thank you Pete. On Thu, May 19, 2016 at 2:44 PM, Pete Kelly <pke...@gmail.com> wrote: > I think this may be the video https://www.youtube.com/watch?v=3XYcQQCWylw > > On 17 May 2016 at 20:41, John Nash <john.nash...@gmail.com> wrote: > >> I saw >> http://www.opensips.org/pub/events/2012-08-07_ClueCon_Chicago/VLAD_PAIU-OpenSIPS-Securing_SIP_Networks.pdf >> . >> >> I would love to watch the video session of this, is there any place I can >> get the video? Tried searching google but did not find. >> >> Regards >> >> John >> >> ___ >> 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 > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Drouting memory usage
OK. will update.Thank you. SIP message validate in the script seemed to have stopped this crash or its not related to that? On Wed, May 18, 2016 at 5:51 PM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > Hi John, > > Thanks for the info, I managed to find the bug and have it fixed: > > https://github.com/OpenSIPS/opensips/commit/4b0fca533cd7be4a45c1381c78f2b37aaba6152b > > Please update from GIT and let me know if you still have the problem. > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 17.05.2016 21:37, John Nash wrote: > > Hello Bogdan, > > The version is Server:: OpenSIPS (2.1.2 (x86_64/linux)) > > Since this is live we do not have detailed debug. Below is what happened > before crash (We had multiple such entries). I figured it might be related > to malformed SIP message so I applied sipmsg_validate() function even for > so called trusted endpoints. > > > May 11 20:38:30 localhost opensips[10315]: ERROR:tm:send_ack: failed to > generate a HBH ACK if key HFs in reply missing > May 11 20:38:30 localhost opensips[10315]: ERROR:tm:reply_received: failed > to send ACK (local=no) > May 11 20:38:46 localhost opensips[10315]: ERROR:tm:send_ack: failed to > generate a HBH ACK if key HFs in reply missing > May 11 20:38:46 localhost opensips[10315]: ERROR:tm:reply_received: failed > to send ACK (local=no) > May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_to: unexpected > char [#015] in status 1: <<"1234 <sip:3437558@x.x.x.x:5060> > <sip:3437558@x.x.x.x:5060>;tag=2878411H96479>> . > May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_from_header: > bad from header > May 11 20:38:51 localhost opensips[10314]: ERROR:dialog:dlg_create_dialog: > bad request or missing FROM hdr :-/ > May 11 20:38:51 localhost opensips[10314]: ERROR:core:skip_name: closing > quote missing in name part of Contact > May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_contacts: > failed to skip name part > May 11 20:38:51 localhost opensips[10314]: ERROR:core:contact_parser: > failed to parse contacts > May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_contact: > failed to parse contact > May 11 20:38:51 localhost opensips[10314]: > ERROR:topology_hiding:topo_no_dlg_encode_contact: bad Contact HDR > May 11 20:38:51 localhost opensips[10314]: ERROR:core:skip_name: closing > quote missing in name part of Contact > May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_contacts: > failed to skip name part > May 11 20:38:51 localhost opensips[10314]: ERROR:core:contact_parser: > failed to parse contacts > May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_contact: > failed to parse contact > May 11 20:38:51 localhost opensips[10314]: > ERROR:topology_hiding:build_encoded_contact_suffix: bad Contact HDR > May 11 20:38:51 localhost opensips[10314]: > CRITICAL:topology_hiding:build_encoded_contact_suffix: #012>>> pkg_free() > on non-pkg ptr 0x18 - aborting!#012#012It seems you have hit a programming > bug.#012Please help us make OpenSIPS better by reporting it at > https://github.com/OpenSIPS/opensips/issues#012 > > > On Tue, May 17, 2016 at 9:35 PM, Bogdan-Andrei Iancu < > <bog...@opensips.org>bog...@opensips.org> wrote: > >> Hi Nash, >> >> What version of OpenSIPS are you using ? also, before that CRITICAL >> message, do you see any other error messages in the logs ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 12.05.2016 08:46, John Nash wrote: >> >> Actually crash happened shortly after we uploaded 11000 codes but looks >> like it is not related to drouting. I see following message >> >> CRITICAL:topology_hiding:build_encoded_contact_suffix: #012>>> pkg_free() >> on non-pkg ptr 0x18 - aborting!#012#012It seems you have hit a programming >> bug.#012Please help us make OpenSIPS better by reporting it at >> <https://github.com/OpenSIPS/opensips/issues#012> >> https://github.com/OpenSIPS/opensips/issues#012 >> >> In log file I see following messages time to time >> ERROR:core:pv_get_contact_body: failed to parse contact hdr >> >> On Wed, May 11, 2016 at 11:29 PM, John Nash < <john.nash...@gmail.com> >> john.nash...@gmail.com> wrote: >> >>> I have been using drouting module with just 200 entries from 8 months >>> yesterday we had need of adding around 11000 entries in rules table but >>> after that opensips started to crash. I am currently using -m 2048 -M 1024 >>> isn't it enough memory? >>> >>> How can I anticipate memory usage? >>> >>> John >>> >> >> >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://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] Security presentation on opensips
I saw http://www.opensips.org/pub/events/2012-08-07_ClueCon_Chicago/VLAD_PAIU-OpenSIPS-Securing_SIP_Networks.pdf . I would love to watch the video session of this, is there any place I can get the video? Tried searching google but did not find. Regards John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Drouting memory usage
Hello Bogdan, The version is Server:: OpenSIPS (2.1.2 (x86_64/linux)) Since this is live we do not have detailed debug. Below is what happened before crash (We had multiple such entries). I figured it might be related to malformed SIP message so I applied sipmsg_validate() function even for so called trusted endpoints. May 11 20:38:30 localhost opensips[10315]: ERROR:tm:send_ack: failed to generate a HBH ACK if key HFs in reply missing May 11 20:38:30 localhost opensips[10315]: ERROR:tm:reply_received: failed to send ACK (local=no) May 11 20:38:46 localhost opensips[10315]: ERROR:tm:send_ack: failed to generate a HBH ACK if key HFs in reply missing May 11 20:38:46 localhost opensips[10315]: ERROR:tm:reply_received: failed to send ACK (local=no) May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_to: unexpected char [#015] in status 1: <<"1234 <sip:3437558@x.x.x.x:5060>;tag=2878411H96479>> . May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_from_header: bad from header May 11 20:38:51 localhost opensips[10314]: ERROR:dialog:dlg_create_dialog: bad request or missing FROM hdr :-/ May 11 20:38:51 localhost opensips[10314]: ERROR:core:skip_name: closing quote missing in name part of Contact May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_contacts: failed to skip name part May 11 20:38:51 localhost opensips[10314]: ERROR:core:contact_parser: failed to parse contacts May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_contact: failed to parse contact May 11 20:38:51 localhost opensips[10314]: ERROR:topology_hiding:topo_no_dlg_encode_contact: bad Contact HDR May 11 20:38:51 localhost opensips[10314]: ERROR:core:skip_name: closing quote missing in name part of Contact May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_contacts: failed to skip name part May 11 20:38:51 localhost opensips[10314]: ERROR:core:contact_parser: failed to parse contacts May 11 20:38:51 localhost opensips[10314]: ERROR:core:parse_contact: failed to parse contact May 11 20:38:51 localhost opensips[10314]: ERROR:topology_hiding:build_encoded_contact_suffix: bad Contact HDR May 11 20:38:51 localhost opensips[10314]: CRITICAL:topology_hiding:build_encoded_contact_suffix: #012>>> pkg_free() on non-pkg ptr 0x18 - aborting!#012#012It seems you have hit a programming bug.#012Please help us make OpenSIPS better by reporting it at https://github.com/OpenSIPS/opensips/issues#012 On Tue, May 17, 2016 at 9:35 PM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > Hi Nash, > > What version of OpenSIPS are you using ? also, before that CRITICAL > message, do you see any other error messages in the logs ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 12.05.2016 08:46, John Nash wrote: > > Actually crash happened shortly after we uploaded 11000 codes but looks > like it is not related to drouting. I see following message > > CRITICAL:topology_hiding:build_encoded_contact_suffix: #012>>> pkg_free() > on non-pkg ptr 0x18 - aborting!#012#012It seems you have hit a programming > bug.#012Please help us make OpenSIPS better by reporting it at > https://github.com/OpenSIPS/opensips/issues#012 > > In log file I see following messages time to time > ERROR:core:pv_get_contact_body: failed to parse contact hdr > > On Wed, May 11, 2016 at 11:29 PM, John Nash <john.nash...@gmail.com> > wrote: > >> I have been using drouting module with just 200 entries from 8 months >> yesterday we had need of adding around 11000 entries in rules table but >> after that opensips started to crash. I am currently using -m 2048 -M 1024 >> isn't it enough memory? >> >> How can I anticipate memory usage? >> >> John >> > > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://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] Drouting memory usage
Actually crash happened shortly after we uploaded 11000 codes but looks like it is not related to drouting. I see following message CRITICAL:topology_hiding:build_encoded_contact_suffix: #012>>> pkg_free() on non-pkg ptr 0x18 - aborting!#012#012It seems you have hit a programming bug.#012Please help us make OpenSIPS better by reporting it at https://github.com/OpenSIPS/opensips/issues#012 In log file I see following messages time to time ERROR:core:pv_get_contact_body: failed to parse contact hdr On Wed, May 11, 2016 at 11:29 PM, John Nash <john.nash...@gmail.com> wrote: > I have been using drouting module with just 200 entries from 8 months > yesterday we had need of adding around 11000 entries in rules table but > after that opensips started to crash. I am currently using -m 2048 -M 1024 > isn't it enough memory? > > How can I anticipate memory usage? > > John > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Drouting memory usage
I have been using drouting module with just 200 entries from 8 months yesterday we had need of adding around 11000 entries in rules table but after that opensips started to crash. I am currently using -m 2048 -M 1024 isn't it enough memory? How can I anticipate memory usage? John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] M4 config generation issue
Yeah changing pass crossed my mind too...:-). I was about to do that but i luckily found this command changecom(`/*', `*/'). This changes comments from "#" to C type comments (For m4 parser). It solved my problem. On Thu, May 5, 2016 at 10:20 PM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > Hi John, > > either dig into M4 secrets , either simply change your password to avoid > the hash char :) > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 05.05.2016 18:20, John Nash wrote: > > I am trying to generate .cfg file with the help of m4. I have following > line in my opensips.cfg.m4 > DB_USER:DB_PASS@DB_IP/DB_NAME > > In defines.m4 I have corresponding values. > > The issue is my DB_PASS contains "#" as one of the character and because > of that any word after DB_PASS is not being replaced (I guess m4 ignores > words after #) > > Any way out? > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://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] M4 config generation issue
I am trying to generate .cfg file with the help of m4. I have following line in my opensips.cfg.m4 DB_USER:DB_PASS@DB_IP/DB_NAME In defines.m4 I have corresponding values. The issue is my DB_PASS contains "#" as one of the character and because of that any word after DB_PASS is not being replaced (I guess m4 ignores words after #) Any way out? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Selective logging
OK thank you.
On Fri, Apr 29, 2016 at 9:59 PM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:
> You can force whatever log level you want via xlog :
> http://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc56
>
> Use L_ALERT to get them all the time :)
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 29.04.2016 19:26, John Nash wrote:
>
> OK. Any way to use Xlog to print DBG messages even if script has debug
> level as ERR ?
>
> On Fri, Apr 29, 2016 at 9:29 PM, Bogdan-Andrei Iancu <
> <bog...@opensips.org>bog...@opensips.org> wrote:
>
>> Hi Nash,
>>
>> You can do a small route in your script for logging to combine filtering
>> (maybe based on src address check or dialplan for usernames or acls, etc)
>> and xlog() :
>>
>> route[my_xlog]
>> {
>> if ( check_source_address("10") )
>> xlog("DBG: $si: $param(1) \n");
>> }
>>
>> And call it as:
>>
>> route(my_xlog,"this is just a simple log");
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 29.04.2016 14:22, John Nash wrote:
>>
>> Is there any way to log messages (Custom messages and SIP trace) from
>> script for a given parameter say IP or ruri.
>>
>> A crude way can be to store say user in local cache and match with the
>> user in script and log else pass but ..
>>
>> 1- I am not sure if any other smart way to do it
>> 2- How can I dunp SIP messages
>>
>>
>>
>>
>> ___
>> Users mailing
>> listUsers@lists.opensips.orghttp://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] Selective logging
Is there any way to log messages (Custom messages and SIP trace) from script for a given parameter say IP or ruri. A crude way can be to store say user in local cache and match with the user in script and log else pass but .. 1- I am not sure if any other smart way to do it 2- How can I dunp SIP messages ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout
Hello Bogdan, The confusion was we received request timeout in two cases .. 1- Gateway did not respond at all not even a trying. (We may take action to disable gateway if too many of such cases) 2- Gateway sent trying and ringing but B party did not answer the call and it timed out. (This is no fault of gateway) This was solved for me by checking which timer expired (fr_timer or fr_inv_timer) I was able to solve it by using your old post. Regards John On Sat, Apr 23, 2016 at 1:52 PM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > Hi John, > > A Request Terminated is generated in response of a received CANCEL > request. In the scenario of a local timeout, there is not received cancel - > it is a timeout event. I do not see any confusion in the reason string in > CDR. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 19.04.2016 17:49, John Nash wrote: > > Ok got it thanks. I also noticed that transactions cancelled because of > fr_inv_timeout , CDR records as "Request timeout". It is quite confusion, > shouldnt it be "Request Terminated"? or I am doing something wrong > > On Tue, Apr 19, 2016 at 6:46 PM, Julian Santer <julian.san...@rolmail.net> > wrote: > >> Hi John, >> >> the commit was: >> >> commit 8133656de9503a122a72c0f80d11eff975bc43f1 >> Author: Bogdan-Andrei Iancu <bog...@opensips.org> >> Date: Thu Feb 11 14:58:41 2016 +0200 >> >> Fix proper callig in local cancels with TH. >> >> Extend the coverage of the preocessing context and TM context over >> the cancel_branch() function (in the timeout handler) so the TH callbacks >> can reach back the dialog and do the TH related changes. >> Reported by Julian Santer on mailing list. >> >> Kind regards, >> Julian Santer >> Raiffeisen OnLine >> >> Am 18.04.2016 um 22:35 schrieb John Nash: >> >>> which revision this was fixed?...I am also using OpenSips 2.1.2 and want >>> to update only this change for the time being (2.2 has many changes) >>> >>> On Thu, Feb 11, 2016 at 7:19 PM, Julian Santer < >>> <julian.san...@rolmail.net>julian.san...@rolmail.net >> julian.san...@rolmail.net>> wrote: >>> >>> Bogdan, >>> >>> we tried now the latest GIT release and it works like a charm ;-) >>> Thank you for quick fix. >>> >>> Kind regards, >>> Julian Santer >>> Raiffeisen OnLine >>> >>> Am 11.02.2016 um 14:02 schrieb Bogdan-Andrei Iancu: >>> >>> Julian, >>> >>> Please update from GIT repo and give it a new try. It should >>> work now (at least it does for me). >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> On 11.02.2016 12:07, Julian Santer wrote: >>> >>> Hi Bogdan, >>> >>> thank you for your time. If you need further informations >>> (config files etc.) let me know. >>> >>> Kind regards, >>> Julian Santer >>> Raiffeisen OnLine >>> >>> Am 11.02.2016 um 10:26 schrieb Bogdan-Andrei Iancu: >>> >>> Hi Julian, >>> >>> I will have to test this and come back to you. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> >>> On 10.02.2016 17:45, Julian Santer wrote: >>> >>> Hi guys, >>> >>> we seem to got the same issue like John Nash on >>> 2015/08/12. >>> We use OpenSips 2.1.2 with the latest revision from >>> git repo. >>> >>> Like John we are not sure if it is a bug or our >>> mistake ;-) >>> >>> We are using topology hiding and the Call ID in the >>> CANCEL, generated by the TM module, is not the same, as the call ID in the >>> initial INVITE. >>> >>> The call flow looks like: >>> PSTN carrier -> gw-carrier (topo hiding) -> core >>> (topo hiding) -> g
[OpenSIPS-Users] Failure cause code in case of transaction timeout
I am using fr_inv_timer and fr_timer and logging failed transactions, but in both cases I get request timeout. Can I control this somehow so that I log "Time out" only in case fr_timer expires and record something else in case fr_inv_timer? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout
Ok got it thanks. I also noticed that transactions cancelled because of fr_inv_timeout , CDR records as "Request timeout". It is quite confusion, shouldnt it be "Request Terminated"? or I am doing something wrong On Tue, Apr 19, 2016 at 6:46 PM, Julian Santer <julian.san...@rolmail.net> wrote: > Hi John, > > the commit was: > > commit 8133656de9503a122a72c0f80d11eff975bc43f1 > Author: Bogdan-Andrei Iancu <bog...@opensips.org> > Date: Thu Feb 11 14:58:41 2016 +0200 > > Fix proper callig in local cancels with TH. > > Extend the coverage of the preocessing context and TM context over the > cancel_branch() function (in the timeout handler) so the TH callbacks can > reach back the dialog and do the TH related changes. > Reported by Julian Santer on mailing list. > > Kind regards, > Julian Santer > Raiffeisen OnLine > > Am 18.04.2016 um 22:35 schrieb John Nash: > >> which revision this was fixed?...I am also using OpenSips 2.1.2 and want >> to update only this change for the time being (2.2 has many changes) >> >> On Thu, Feb 11, 2016 at 7:19 PM, Julian Santer <julian.san...@rolmail.net >> <mailto:julian.san...@rolmail.net>> wrote: >> >> Bogdan, >> >> we tried now the latest GIT release and it works like a charm ;-) >> Thank you for quick fix. >> >> Kind regards, >> Julian Santer >> Raiffeisen OnLine >> >> Am 11.02.2016 um 14:02 schrieb Bogdan-Andrei Iancu: >> >> Julian, >> >> Please update from GIT repo and give it a new try. It should work >> now (at least it does for me). >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 11.02.2016 12:07, Julian Santer wrote: >> >> Hi Bogdan, >> >> thank you for your time. If you need further informations >> (config files etc.) let me know. >> >> Kind regards, >> Julian Santer >> Raiffeisen OnLine >> >> Am 11.02.2016 um 10:26 schrieb Bogdan-Andrei Iancu: >> >> Hi Julian, >> >> I will have to test this and come back to you. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 10.02.2016 17:45, Julian Santer wrote: >> >> Hi guys, >> >> we seem to got the same issue like John Nash on >> 2015/08/12. >> We use OpenSips 2.1.2 with the latest revision from >> git repo. >> >> Like John we are not sure if it is a bug or our >> mistake ;-) >> >> We are using topology hiding and the Call ID in the >> CANCEL, generated by the TM module, is not the same, as the call ID in the >> initial INVITE. >> >> The call flow looks like: >> PSTN carrier -> gw-carrier (topo hiding) -> core >> (topo hiding) -> gw-consumer (topo-hiding) -> UAC consumer >> >> The CANCEL generated by the TM module of the core, >> sended to the gw-consumer is rejected by the gw-consumer. >> >> The CANCEL starts on the core. So let me show you >> 1) the initial INVITE, which the core receives from >> the gw-carrier (Call-ID: GW-CARRIER) >> 2) the initial INVITE, which the core and sends to >> the gw-consumer (Call-ID: Core) >> 3) the CANCEL generated by the core after >> $T_fr_inv_timeout (Call-ID: GW-CARRIER) >> >> 1) >> INVITE sip:12345@IP_CORE SIP/2.0 >> Via: SIP/2.0/UDP >> IP_GW-CARRIER:5060;branch=z9hG4bK6aa2.7710f555.0 >> From: <sip:+396789@domain>;tag=E3AE5C5C-1A42 >> To: <sip:12345@domain> >> Call-ID: >> GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE >> CSeq: 101 INVITE >> Max-Forwards: 8 >> Remote-Party-ID: <sip:+396789@IP_CARRIER >> >;party=calling;screen=yes;privacy=off >> Contact: <sip:+396789@IP_GW
Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout
which revision this was fixed?...I am also using OpenSips 2.1.2 and want to update only this change for the time being (2.2 has many changes) On Thu, Feb 11, 2016 at 7:19 PM, Julian Santer <julian.san...@rolmail.net> wrote: > Bogdan, > > we tried now the latest GIT release and it works like a charm ;-) > Thank you for quick fix. > > Kind regards, > Julian Santer > Raiffeisen OnLine > > Am 11.02.2016 um 14:02 schrieb Bogdan-Andrei Iancu: > >> Julian, >> >> Please update from GIT repo and give it a new try. It should work now (at >> least it does for me). >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> On 11.02.2016 12:07, Julian Santer wrote: >> >>> Hi Bogdan, >>> >>> thank you for your time. If you need further informations (config files >>> etc.) let me know. >>> >>> Kind regards, >>> Julian Santer >>> Raiffeisen OnLine >>> >>> Am 11.02.2016 um 10:26 schrieb Bogdan-Andrei Iancu: >>> >>>> Hi Julian, >>>> >>>> I will have to test this and come back to you. >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> >>>> On 10.02.2016 17:45, Julian Santer wrote: >>>> >>>>> Hi guys, >>>>> >>>>> we seem to got the same issue like John Nash on 2015/08/12. >>>>> We use OpenSips 2.1.2 with the latest revision from git repo. >>>>> >>>>> Like John we are not sure if it is a bug or our mistake ;-) >>>>> >>>>> We are using topology hiding and the Call ID in the CANCEL, generated >>>>> by the TM module, is not the same, as the call ID in the initial INVITE. >>>>> >>>>> The call flow looks like: >>>>> PSTN carrier -> gw-carrier (topo hiding) -> core (topo hiding) -> >>>>> gw-consumer (topo-hiding) -> UAC consumer >>>>> >>>>> The CANCEL generated by the TM module of the core, sended to the >>>>> gw-consumer is rejected by the gw-consumer. >>>>> >>>>> The CANCEL starts on the core. So let me show you >>>>> 1) the initial INVITE, which the core receives from the gw-carrier >>>>> (Call-ID: GW-CARRIER) >>>>> 2) the initial INVITE, which the core and sends to the gw-consumer >>>>> (Call-ID: Core) >>>>> 3) the CANCEL generated by the core after $T_fr_inv_timeout (Call-ID: >>>>> GW-CARRIER) >>>>> >>>>> 1) >>>>> INVITE sip:12345@IP_CORE SIP/2.0 >>>>> Via: SIP/2.0/UDP IP_GW-CARRIER:5060;branch=z9hG4bK6aa2.7710f555.0 >>>>> From: <sip:+396789@domain>;tag=E3AE5C5C-1A42 >>>>> To: <sip:12345@domain> >>>>> Call-ID: >>>>> GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE >>>>> CSeq: 101 INVITE >>>>> Max-Forwards: 8 >>>>> Remote-Party-ID: <sip:+396789@IP_CARRIER >>>>> >;party=calling;screen=yes;privacy=off >>>>> Contact: <sip:+396789@IP_GW-CARRIER;rdlg=3db.94186637> >>>>> Expires: 180 >>>>> Content-Type: application/sdp >>>>> Content-Length: 474 >>>>> sdp ... >>>>> >>>>> 2) >>>>> INVITE sip:12345@IP_UAC:PORT_UAC SIP/2.0 >>>>> Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0 >>>>> Route: <sip:IP_GW-CONSUMER;lr> >>>>> From: <sip:+396789@domain>;tag=E3AE5C5C-1A42 >>>>> To: <sip:12345@domain> >>>>> Call-ID: >>>>> Core_ExwGCwAcKhgvdAgnFg58LwQGAXUOXSAzC3A1JFB/FCcWCWFzGAQkXHInPFQGDzYYI3oPCCAMahZeEy11JywlCVEG >>>>> CSeq: 101 INVITE >>>>> Max-Forwards: 7 >>>>> Remote-Party-ID: <sip:+396789@IP_CARRIER >>>>> >;party=calling;screen=yes;privacy=off >>>>> Contact: <sip:+396789@IP_CORE;rdlg=28e.bad6c124> >>>>> Expires: 180 >>>>> Content-Type: application/sdp >>>>> Content-Length: 426 >>>>> sdp ... >>>>> >>>>> 3) >>>>> CANCEL sip:12345@IP_UAC:PORT_UAC SIP/2.0 >>>>> Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0 >>>>> From: <sip:+396789@domain>;tag=E3AE5C5C-1A42 >>>>> Call-ID: >>>>> GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE >>>>> To: <sip:12345@domain> >>>>> CSeq: 101 CANCEL >>>>> Max-Forwards: 70 >>>>> Route: <sip:IP_GW-CONSUMER;lr> >>>>> Reason: SIP;cause=480;text="NO_ANSWER" >>>>> User-Agent: OpenSIPS (2.1.2 (x86_64/linux)) >>>>> Content-Length: 0 >>>>> >>>>> Kind regards, >>>>> Julian Santer >>>>> Raiffeisen OnLine >>>>> >>>>> >>>>> ___ >>>>> 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 >>> >>> >> >> > > > ___ > 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] Dialogs stay in memory until restart in state 5
I am not able to find the commit can you please point me? On Wed, Jan 13, 2016 at 9:07 AM, John Nash <john.nash...@gmail.com> wrote: > OK thank you I will try to find and patch. > > On Tue, Jan 12, 2016 at 10:58 PM, Răzvan Crainea <raz...@opensips.org> > wrote: > >> Hi, John! >> >> This issue was fixed in newer versions of OpenSIPS, and the fix will be >> part of OpenSIPS 2.1.2. >> >> Best regards, >> Răzvan >> >> >> On 01/12/2016 03:19 PM, John Nash wrote: >> >> I am using OpenSIPS (2.1.1 (x86_64/linux)) with dialogs module and >> topology hiding modules. >> I am not saving dialogs in database. >> >> After I run it for few hours and stop traffic i see hundreds of dialogs >> using fifo command which wont be deleted from memory until I restart. I see >> dialogs like .. >> dialog:: hash=4094:1281471850 >> state:: 5 >> user_flags:: 0 >> timestart:: 0 >> timeout:: 0 >> callid:: 8abe7f9e-2cd5-1234-84b0-d4ae52bce047 >> from_uri:: sip:4117834663@127.0.0.1 >> to_uri:: sip:119603452947@127.0.0.1:5060 >> >> I saw couple of old posts and tried using "Pp" parameter in create_dialog >> but it does not seem to be of any effect. >> >> In logs I also see messages like "ERROR:dialog:push_reply_in_dialog: >> [487] reply in dlg state [2]: missing TAG param in TO hdr" which I am not >> sure are related to my problem. >> >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> -- >> Răzvan Crainea >> OpenSIPS Core Developerhttp://www.opensips-solutions.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] Cache design
Yes I am also doing that with local cache but I was confused with sql_cacher and with mongo db also it can be done?...I was just wondering what are these for? On Wed, Jan 13, 2016 at 1:40 AM, <jar...@unixc.org> wrote: > John, > > I do something similar for caching auth user id and passwords using > cachedb_redis/cachedb_local and db_postgres. > > My basic logic first checks cache (cache_fetch) to see if the key exists. > If it does I will pv_proxy_authorize using the value stored in the > populated avp from the fetch. If the cache does not exist, I will query > Postgres (avp_db_query) and store (cache_store) the value in cache with a > lifetime so that it will automatically repopulate at a later time. The > only time I may adjust the cache out of band would be if the user modifies > their SIP password through the portal which causes the system to delete the > stale key/value from the cachedb store. > > This may or may not address your concern, but I hope it helps. > > Jarrod > > > On Jan 12, 2016, at 1:54 PM, John Nash <john.nash...@gmail.com> wrote: > > > > I am using local cache db module in order to keep user id and password > in memory and now plan to keep some other data in memory too and that I > want to keep in some centralized cache. > > > > It will be like Opensips1, Opensips2 <-> Cache server > <-> Postgresql > > > > I also need to periodically update cache (using some mi commands may be) > > > > I saw many of the modules related to cache in opensips (Like sql_cacher, > cachedb_* , DB_CACHEDB) and I am confused on what should I use for my need > and how these are all related. > > > > Any suggestions? > > ___ > > 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 > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialogs stay in memory until restart in state 5
OK thank you I will try to find and patch. On Tue, Jan 12, 2016 at 10:58 PM, Răzvan Crainea <raz...@opensips.org> wrote: > Hi, John! > > This issue was fixed in newer versions of OpenSIPS, and the fix will be > part of OpenSIPS 2.1.2. > > Best regards, > Răzvan > > > On 01/12/2016 03:19 PM, John Nash wrote: > > I am using OpenSIPS (2.1.1 (x86_64/linux)) with dialogs module and > topology hiding modules. > I am not saving dialogs in database. > > After I run it for few hours and stop traffic i see hundreds of dialogs > using fifo command which wont be deleted from memory until I restart. I see > dialogs like .. > dialog:: hash=4094:1281471850 > state:: 5 > user_flags:: 0 > timestart:: 0 > timeout:: 0 > callid:: 8abe7f9e-2cd5-1234-84b0-d4ae52bce047 > from_uri:: sip:4117834663@127.0.0.1 > to_uri:: sip:119603452947@127.0.0.1:5060 > > I saw couple of old posts and tried using "Pp" parameter in create_dialog > but it does not seem to be of any effect. > > In logs I also see messages like "ERROR:dialog:push_reply_in_dialog: [487] > reply in dlg state [2]: missing TAG param in TO hdr" which I am not sure > are related to my problem. > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- > Răzvan Crainea > OpenSIPS Core Developerhttp://www.opensips-solutions.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
[OpenSIPS-Users] Dialogs stay in memory until restart in state 5
I am using OpenSIPS (2.1.1 (x86_64/linux)) with dialogs module and topology hiding modules. I am not saving dialogs in database. After I run it for few hours and stop traffic i see hundreds of dialogs using fifo command which wont be deleted from memory until I restart. I see dialogs like .. dialog:: hash=4094:1281471850 state:: 5 user_flags:: 0 timestart:: 0 timeout:: 0 callid:: 8abe7f9e-2cd5-1234-84b0-d4ae52bce047 from_uri:: sip:4117834663@127.0.0.1 to_uri:: sip:119603452947@127.0.0.1:5060 I saw couple of old posts and tried using "Pp" parameter in create_dialog but it does not seem to be of any effect. In logs I also see messages like "ERROR:dialog:push_reply_in_dialog: [487] reply in dlg state [2]: missing TAG param in TO hdr" which I am not sure are related to my problem. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Cache design
I am using local cache db module in order to keep user id and password in memory and now plan to keep some other data in memory too and that I want to keep in some centralized cache. It will be like Opensips1, Opensips2 <-> Cache server <-> Postgresql I also need to periodically update cache (using some mi commands may be) I saw many of the modules related to cache in opensips (Like sql_cacher, cachedb_* , DB_CACHEDB) and I am confused on what should I use for my need and how these are all related. Any suggestions? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] In Dialog Options message to check if call is alive
I am using Create Dialog with "Pp" as parameter but I have one doubt, what
if the either side of Opensips is another proxy or SBC which has following
code at the start (Before loose route or topology hiding check)
if(((is_method("NOTIFY") && $hdr(Event) =~ "keep-alive") ||
is_method("OPTIONS")))
{
sl_send_reply("200", "Alive");
exit;
}
Then i will get a 200 OK for every option message whether it is generated
by dialog module or drouting module as gateway ping?...So I fear if in
dialog ping is a reliable way to check if call is alive?
___
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[OpenSIPS-Users] Tight matching of dialog failed
I get following warning in log Dec 1 08:40:14 [14516] WARNING:dialog:dlg_onroute: tight matching failed for BYE with callid='[%!< to}'/36, ftag='9r60aB1Q77tUj'/13, ttag='PROXY1448958761787to'/20 and direction=0 Dec 1 08:40:14 [14516] WARNING:dialog:dlg_onroute: dialog identification elements are callid='DLGCH_FFVEUltVXEVZBAoTXFlQW19bWE0MAENIRAIKWFgURQYLRlcR'/54, caller tag='9r60aB1Q77tUj'/13, callee tag='PROXY1448958761787to'/20 Strange thing is that when i see the dump in wireshark, I can see call-id is correct it is not what is printed in above message. My scenario is One opensips sending call to second opensips (both using topology hiding). I pasted log from second opensips. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Security related
I have couple of things i need your valuable inputs I have already seen some articles and slides but some questions remain... 1- AVP db queries do we need to escape parameters or its taken care of by module internally. 2- How can I secure opensipsctl and mi_datagram as that is gateway to my opensips. 3- I need to use exec module to run some opensipsctl commands, If I understand correctly, if i am running opensips as root someone can run any command on my box?...On the other hand if I run opensips on some non privileged user can I still run exec? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Drouting stripping and adding prefix
Thank you. It works that way for me. On Wed, Nov 18, 2015 at 3:45 PM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > Hi John, > > Simply redefine the gateway multiple times, same IP, but different ID, > strip and prefix > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 17.11.2015 08:34, John Nash wrote: > > I am using Drouting module and quite happy with it. I have some new > requirement where I need to be able to strip some digits from "incoming" > number and add some prefix at dr_rule level. I know I can do that at > dr_gateway wise but as per my new requirement I need to use same gateway > for routing for multiple dr_rules with different strip and prefix. > > Is there any way to do it?...Do I need to look into some other module like > dialing plan? or may be there is some way of doing it in script with the > help of "attr" field of dr_routing? > > Regards, > > John > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://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] Drouting stripping and adding prefix
I am using Drouting module and quite happy with it. I have some new requirement where I need to be able to strip some digits from "incoming" number and add some prefix at dr_rule level. I know I can do that at dr_gateway wise but as per my new requirement I need to use same gateway for routing for multiple dr_rules with different strip and prefix. Is there any way to do it?...Do I need to look into some other module like dialing plan? or may be there is some way of doing it in script with the help of "attr" field of dr_routing? Regards, John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Ack without To tag
Anyone has thoughts on this?.If i use record routing instead of topology hiding would it help?. On Wed, Jul 15, 2015 at 10:47 AM, John Nash <john.nash...@gmail.com> wrote: > Dear Vlad, > > Do you need any more information? Like debug log or complete wireshark > pcap? > > John > > On Wed, Jun 24, 2015 at 10:06 PM, John Nash <john.nash...@gmail.com> > wrote: > >> I am using opensips 2.1 with topology_hiding module. I have an issue only >> with one SIP endpoint. This endpoint sends Ack message (after 200 OK to >> Invite) without any to tag because of that it is not matching with In >> dialog request section. >> >> Can a UA send ACK without to tag?...If yes any way I can match it with >> ongoing dialog? >> >> John >> > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Call to registered user -- Caller id search
OK..I also looked at alias module. I am really fascinated by cache any way
to use alias db with cache?
On Wed, Sep 23, 2015 at 7:23 PM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:
> Hi Nash,
>
> If the registration of user2 was handled via save("location"), you can
> route the call to actual IP of user2 via lookup("location").
>
> To translate between the 2 number and User2 (before the lookup),
> use the aliases module.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 17.09.2015 17:12, John Nash wrote:
>
> I am trying to test a use case in which I need to send call to a
> registered user based on its caller ID.
>
> User1
> Username = 1001
> Carrier id = 11
>
> User2
> Username = 1002
> Carrier id = 2
>
> Both are registered to my opensips. Now if User1 calls number 2 ,
> I want opensips to try to reach User2 if its online.
>
> Is this possible?
>
>
> ___
> Users mailing
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
___
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Re: [OpenSIPS-Users] Call to registered user -- Caller id search
I think I can use local_cache and store caller id as key and sip user as
value. Right?
On Thu, Sep 24, 2015 at 8:26 AM, John Nash <john.nash...@gmail.com> wrote:
> OK..I also looked at alias module. I am really fascinated by cache any way
> to use alias db with cache?
>
> On Wed, Sep 23, 2015 at 7:23 PM, Bogdan-Andrei Iancu <bog...@opensips.org>
> wrote:
>
>> Hi Nash,
>>
>> If the registration of user2 was handled via save("location"), you can
>> route the call to actual IP of user2 via lookup("location").
>>
>> To translate between the 2 number and User2 (before the lookup),
>> use the aliases module.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 17.09.2015 17:12, John Nash wrote:
>>
>> I am trying to test a use case in which I need to send call to a
>> registered user based on its caller ID.
>>
>> User1
>> Username = 1001
>> Carrier id = 11
>>
>> User2
>> Username = 1002
>> Carrier id = 2
>>
>> Both are registered to my opensips. Now if User1 calls number 2 ,
>> I want opensips to try to reach User2 if its online.
>>
>> Is this possible?
>>
>>
>> ___
>> Users mailing
>> listUsers@lists.opensips.orghttp://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] Call to registered user -- Caller id search
I am trying to test a use case in which I need to send call to a registered user based on its caller ID. User1 Username = 1001 Carrier id = 11 User2 Username = 1002 Carrier id = 2 Both are registered to my opensips. Now if User1 calls number 2 , I want opensips to try to reach User2 if its online. Is this possible? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm
Yes it is the original A leg Invite. On Thu, Aug 13, 2015 at 2:45 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi John, Is the callid in CANCEL the original one from the incoming INVITE ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 12.08.2015 09:52, John Nash wrote: I am not sure if its some bug or my mistake. I am using topology hiding module (opensips 2.1 version) and I have noticed that Call-id in Cancel message is different than Invite sent to gateway. Invite is sent to gateway and we get session progress but call is not picked up, as per fr_timer opensips triggers cancel but call ID in cancel is not same as sent in Invite INVITE sip:11025@2.2.2.2:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:9096;branch=z9hG4bKe5d5.fca1aeb4.0 Max-Forwards: 26 From: 786786 sip:786786@1.1.1.1:9096;tag=2B6F7HZaNH02F To: sip:11025@2.2.2.2:5060 Call-ID: DLGCH_LkhSV2VxNGNiEgQMMGRhYXxDSF9rcDN+K0QEC2Z7M2ouFFFd CSeq: 79309702 INVITE Contact: sip:sbc@1.1.1.1:9096;did=775.84850bf2 sip:sbc@1.1.1.1:9096;did=775.84850bf2 User-Agent: Opensips Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 258 v=0 o=- 22469580 22469580 IN IP4 1.1.1.1 s=Softphone c=IN IP4 1.1.1.1 t=0 0 m=audio 32706 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=rtcp:32707 CANCEL sip:11025@2.2.2.2:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:9096;branch=z9hG4bKe5d5.fca1aeb4.0 From: 786786 sip:786786@1.1.1.1:9096;tag=2B6F7HZaNH02F Call-ID: a87968d0-babc-1233-189c-d4ae52c9ad43 To: sip:11025@2.2.2.2:5060 CSeq: 79309702 CANCEL Max-Forwards: 70 Reason: SIP;cause=480;text=NO_ANSWER User-Agent: Opensips Content-Length: 0 ___ Users mailing listUsers@lists.opensips.orghttp://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] Call-id issue in Cancel message generated by tm
I am not sure if its some bug or my mistake. I am using topology hiding module (opensips 2.1 version) and I have noticed that Call-id in Cancel message is different than Invite sent to gateway. Invite is sent to gateway and we get session progress but call is not picked up, as per fr_timer opensips triggers cancel but call ID in cancel is not same as sent in Invite INVITE sip:11025@2.2.2.2:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:9096;branch=z9hG4bKe5d5.fca1aeb4.0 Max-Forwards: 26 From: 786786 sip:786786@1.1.1.1:9096;tag=2B6F7HZaNH02F To: sip:11025@2.2.2.2:5060 Call-ID: DLGCH_LkhSV2VxNGNiEgQMMGRhYXxDSF9rcDN+K0QEC2Z7M2ouFFFd CSeq: 79309702 INVITE Contact: sip:sbc@1.1.1.1:9096;did=775.84850bf2 User-Agent: Opensips Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 258 v=0 o=- 22469580 22469580 IN IP4 1.1.1.1 s=Softphone c=IN IP4 1.1.1.1 t=0 0 m=audio 32706 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=rtcp:32707 CANCEL sip:11025@2.2.2.2:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:9096;branch=z9hG4bKe5d5.fca1aeb4.0 From: 786786 sip:786786@1.1.1.1:9096;tag=2B6F7HZaNH02F Call-ID: a87968d0-babc-1233-189c-d4ae52c9ad43 To: sip:11025@2.2.2.2:5060 CSeq: 79309702 CANCEL Max-Forwards: 70 Reason: SIP;cause=480;text=NO_ANSWER User-Agent: Opensips Content-Length: 0 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Ack without To tag
Dear Vlad, Do you need any more information? Like debug log or complete wireshark pcap? John On Wed, Jun 24, 2015 at 10:06 PM, John Nash john.nash...@gmail.com wrote: I am using opensips 2.1 with topology_hiding module. I have an issue only with one SIP endpoint. This endpoint sends Ack message (after 200 OK to Invite) without any to tag because of that it is not matching with In dialog request section. Can a UA send ACK without to tag?...If yes any way I can match it with ongoing dialog? John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.11 on Centos 6.6, running but not listening?
Hello Frank, I saw your message http://lists.opensips.org/pipermail/users/2015-March/031296.html Did you get any head or tail of this issue?..I also face the same situation where in wireshark I see perfect messages but opensips log shows unable to parse and shows junk characters John On Thu, Mar 26, 2015 at 7:37 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi, First check if opensips actually started : run ps auxw | grep opensips If it didn't , check the logs (messages or syslog) for errors. If you see the process, check with netstat -lnp | grep opensips to see the listening interfaces of OpenSIPS. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 25.03.2015 17:41, bluerain wrote: I am currently running Opensips 1.9 on Debian Whizzy, but as stated on my other thread, I am getting some weird error which opensips would suddenly stop working. So, I am starting a new installation of Opensips 1.11 on Centos 6.6 using the yum install method: 1. I was able to install yum opensips install with no issue, but there are some module it didn't install which I need (e.g. db_unixodbc and db_mysql and httpd) 2. So I went ahead download the source and all the dependency files needed and re-complie the oepnsips to get those .so files 3. After some tweaking of the database table structure and some other stuff, I was able to get opensips 1.11 running without error (on the message.log file) using my 1.9 config file. So I did opensipsctl restart (just to make sure opensips starts) I see: INFO: Restarting OpenSIPS : INFO: stopped INFO: Starting OpenSIPS : INFO: Removing stale PID file /var/run/opensips.pid. INFO: started (pid: 2079) but when I try to use a VoIP device to register to it, it doesn't response. thus I did: sudo netstat -plnt The only think I see that opensips is listening on is port by PID 2080 (I don't see 2079 at all) So what is happening? why no error and it seems opensips running but yet is not listsening on port 5060? I have 2 nic card and eth0 is the one with public IP and the default route where as the eth1 is the one with private IP and no default route. Thus I am sure everything is route out of the eth0 (public IP). When I do simply the command opensips, it would simply return: Listening on udp: xxx.xxx.xxx.xxx [xxx.xxx.xxx.xxx]:5060 Aliases: *: (my FQDN):* So it looks like it think is listening on port 5060 but is not? Any help would be appreciated greatly! Thank you! -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-11-on-Centos-6-6-running-but-not-listening-tp7596152.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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Media-proxy dead air issue
It seems like media proxy flag related issue to me. My guess is from first gateway which fails you get session progress and then it rejects and you get another session progress from second gateway. If you play around with mediaproxy flags (at the time of session progress and 200 OK) it can be solved. On Wed, Jul 1, 2015 at 1:19 AM, pwilli...@turnopen.com wrote: I'm encountering a dead air issue with mediaproxy (version 2.5.2), specifically media-relay application. Seems similar to issue reported by Edwin http://lists.opensips.org/pipermail/users/2015-April/031498.html (no responses) My scenario is that an updated rtp port is sent by my clientSBC after failure of initial gateway, media-relay seems to get the new port via 183, but does not modify the stream correctly. I started media relay with no-fork option and received below. First rtp port negotiated. clientSBC:10568 Updated rtp port requested clientSBC:10570 Expecting this clientSBC:10570 (RTP: clientSBC:10570, RTCP: Unknown) Current Result (old port 10568, still remains) clientSBC:10570 (RTP: clientSBC:10568, RTCP: Unknown) debug: Received new SDP offer mediaproxy.mediacontrol.StreamListenerProtocol starting on 1 mediaproxy.mediacontrol.StreamListenerProtocol starting on 10001 mediaproxy.mediacontrol.StreamListenerProtocol starting on 10002 mediaproxy.mediacontrol.StreamListenerProtocol starting on 10003 debug: Added new stream: (audio) internal-IVR:27980 (RTP: Unknown, RTCP: Unknown) - OpenSIPS:1 - OpenSIPS:10002 - Unknown (RTP: Unknown, RTCP: Unknown) debug: created new session 7cf56d31-8aff-1233-8086-001a4a10fa59: 7185551212@internal-IVR (F366pUF6HNHeg) -- 7325553535@outsideProvider debug: Got traffic information for stream: (audio) internal-IVR:27980 (RTP: Unknown, RTCP: Unknown) - OpenSIPS:1 - OpenSIPS:10002 - Unknown (RTP: clientSBC:10568, RTCP: Unknown) debug: updating existing session 7cf56d31-8aff-1233-8086-001a4a10fa59: 7185551212@internal-IVR (F366pUF6HNHeg) -- 7325553535@outsideProvider debug: Received updated SDP answer debug: Got initial answer from callee for stream: (audio) internal-IVR:27980 (RTP: Unknown, RTCP: Unknown) - OpenSIPS:1 - OpenSIPS:10002 - clientSBC:10570 (RTP: clientSBC:10568, RTCP: Unknown) debug: updating existing session 7cf56d31-8aff-1233-8086-001a4a10fa59: 7185551212@internal-IVR (F366pUF6HNHeg) -- 7325553535@outsideProvider debug: Received updated SDP answer debug: Unchanged stream: (audio) internal-IVR:27980 (RTP: Unknown, RTCP: Unknown) - OpenSIPS:1 - OpenSIPS:10002 - clientSBC:10570 (RTP: clientSBC:10568, RTCP: Unknown) debug: updating existing session 7cf56d31-8aff-1233-8086-001a4a10fa59: 7185551212@internal-IVR (F366pUF6HNHeg) -- 7325553535@outsideProvider debug: Received new SDP offer debug: Found matching existing stream: (audio) internal-IVR:27980 (RTP: Unknown, RTCP: Unknown) - OpenSIPS:1 - OpenSIPS:10002 - clientSBC:10570 (RTP: clientSBC:10568, RTCP: Unknown) debug: updating existing session 7cf56d31-8aff-1233-8086-001a4a10fa59: 7185551212@internal-IVR (F366pUF6HNHeg) -- 7325553535@outsideProvider debug: Received updated SDP answer debug: Unchanged stream: (audio) internal-IVR:27980 (RTP: Unknown, RTCP: Unknown) - OpenSIPS:1 - OpenSIPS:10002 - clientSBC:10570 (RTP: clientSBC:10568, RTCP: Unknown) debug: Got traffic information for stream: (audio) internal-IVR:27980 (RTP: internal-IVR:27980, RTCP: Unknown) - OpenSIPS:1 - OpenSIPS:10002 - clientSBC:10570 (RTP: clientSBC:10568, RTCP: Unknown) debug: removing session 7cf56d31-8aff-1233-8086-001a4a10fa59: 7185551212@internal-IVR (F366pUF6HNHeg) -- 7325553535@outsideProvider (Port 1 Closed) (Port 10001 Closed) (Port 10002 Closed) (Port 10003 Closed) Seems like a bug. Anyone experienced this before... is there a fix.. If no fix, anyone familiar enough with the codebase to point out how to update the port object/attribute for the stream? I'm troubleshooting/modifying code on my own, but wanted to cover all bases. Thanks, Paul W. ___ 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] Ack without To tag
Regards, Vlad Paiu OpenSIPS Developerhttp://www.opensips-solutions.com On 25.06.2015 06:25, John Nash wrote: Let me add some more details which I noticed. As i explained in previous post in the same subject, we receive ACK without to tag from one UA, since it will not match any dialog, I tried to route it to destination with topology_hiding but topology hiding used different branch tag than the 200 OK received (It only replaced .0 with .2). I think this is happening because I am also using drouting module and UAC which use branches. I can post trace also if someone wants to have a look. On Wed, Jun 24, 2015 at 10:06 PM, John Nash john.nash...@gmail.com wrote: I am using opensips 2.1 with topology_hiding module. I have an issue only with one SIP endpoint. This endpoint sends Ack message (after 200 OK to Invite) without any to tag because of that it is not matching with In dialog request section. Can a UA send ACK without to tag?...If yes any way I can match it with ongoing dialog? John ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ 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] topology_hiding function and loose_route etc
Hello Bogdan, This issue is solved. It turned out to be my config issue. I was stripping some headers in initial Invite while in case of Reinvite I passed them as it is and that broke something at gateway. But I am facing a real issue which i cannot get past. I sent another mail by subject Ack without To tag. Once convenient please check that. John On Mon, Jun 29, 2015 at 3:07 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi John, So you have a SIP capture (full call) to show the re-INVITE problem ? Otherwise it is hard to figure out what is going on there. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 15.06.2015 22:08, John Nash wrote: Hello Bogdan, Thank you I will just ignore them. I have one more related issue. I am using uac_replace_from in auto mode along with topology_hiding. In a case when UA sends opensips a REInvite , my end carrier seem to completely ignore the Reinvite. I noticed that From URI in original Invite is different from the one sent in Reinvite (Only change is caller ID) Is there something I should know when mixing topology_hiding function and uac_replace_from? Regards John On Mon, Jun 15, 2015 at 9:57 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi John, The module complains of receiving in Early state a reply without tag param in TO header - something like that is bogus. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 15.06.2015 12:32, John Nash wrote: I have modified my proxy config to support topology_hiding function of dialog module. But I see lot of dialog related errors like .. ERROR:dialog:push_reply_in_dialog: [487] reply in dlg state [2]: missing TAG param in TO hdr ERROR:dialog:w_validate_dialog: null dialog I am just wondering if my configuration is correct. How functions like loose_route(); match_dialog();, Validate_dialog(), fix_route_dialog should be used in production environment to cover all cases. From documentation I find code snippets explaining application for each function but how they all work together in topology_hiding function scenario? PS: I can send my config in case someone needs to have a look. John ___ Users mailing listUsers@lists.opensips.orghttp://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 2.0 -- toplogy_hiding_match logic
Is it required for SIP message to have a to-tag in order to be matched with dialog using topology_hiding_match or match_dialog? In one situation ACK message from UA does not have to-tag but From-tag, call-id and all headers seems to belong to ongoing dialog? Is there a way to match such request with dialog? John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Ack without To tag
Let me add some more details which I noticed. As i explained in previous post in the same subject, we receive ACK without to tag from one UA, since it will not match any dialog, I tried to route it to destination with topology_hiding but topology hiding used different branch tag than the 200 OK received (It only replaced .0 with .2). I think this is happening because I am also using drouting module and UAC which use branches. I can post trace also if someone wants to have a look. On Wed, Jun 24, 2015 at 10:06 PM, John Nash john.nash...@gmail.com wrote: I am using opensips 2.1 with topology_hiding module. I have an issue only with one SIP endpoint. This endpoint sends Ack message (after 200 OK to Invite) without any to tag because of that it is not matching with In dialog request section. Can a UA send ACK without to tag?...If yes any way I can match it with ongoing dialog? John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Ack without To tag
I am using opensips 2.1 with topology_hiding module. I have an issue only with one SIP endpoint. This endpoint sends Ack message (after 200 OK to Invite) without any to tag because of that it is not matching with In dialog request section. Can a UA send ACK without to tag?...If yes any way I can match it with ongoing dialog? John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] topology_hiding function and loose_route etc
I have modified my proxy config to support topology_hiding function of dialog module. But I see lot of dialog related errors like .. ERROR:dialog:push_reply_in_dialog: [487] reply in dlg state [2]: missing TAG param in TO hdr ERROR:dialog:w_validate_dialog: null dialog I am just wondering if my configuration is correct. How functions like loose_route(); match_dialog();, Validate_dialog(), fix_route_dialog should be used in production environment to cover all cases. From documentation I find code snippets explaining application for each function but how they all work together in topology_hiding function scenario? PS: I can send my config in case someone needs to have a look. John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] topology_hiding function and loose_route etc
Hello Bogdan, Thank you I will just ignore them. I have one more related issue. I am using uac_replace_from in auto mode along with topology_hiding. In a case when UA sends opensips a REInvite , my end carrier seem to completely ignore the Reinvite. I noticed that From URI in original Invite is different from the one sent in Reinvite (Only change is caller ID) Is there something I should know when mixing topology_hiding function and uac_replace_from? Regards John On Mon, Jun 15, 2015 at 9:57 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi John, The module complains of receiving in Early state a reply without tag param in TO header - something like that is bogus. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 15.06.2015 12:32, John Nash wrote: I have modified my proxy config to support topology_hiding function of dialog module. But I see lot of dialog related errors like .. ERROR:dialog:push_reply_in_dialog: [487] reply in dlg state [2]: missing TAG param in TO hdr ERROR:dialog:w_validate_dialog: null dialog I am just wondering if my configuration is correct. How functions like loose_route(); match_dialog();, Validate_dialog(), fix_route_dialog should be used in production environment to cover all cases. From documentation I find code snippets explaining application for each function but how they all work together in topology_hiding function scenario? PS: I can send my config in case someone needs to have a look. John ___ Users mailing listUsers@lists.opensips.orghttp://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] Version question
I had started testing 1.X series taken from github master branch couple of months ago (It shows version as Server:: OpenSIPS (1.12.0dev-notls (x86_64/linux)) Now I need to install it in one of the production server and before I do that I want to update to the latest version of 1.X series. Which branch should I use to use most upto date 1.X opensips (As master branch now is 2.1) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Version question
OK. Thank you. I have one more related question. If I use 2.1 version (I understand there may be some bugs) and use it with 1.X scripts (I mean without using async features in script for now) should it all work in general?...Like all modules? On Sun, Mar 8, 2015 at 9:52 PM, Liviu Chircu li...@opensips.org wrote: Hello John, 1.11.3 LTS seems to be what you're looking for [1]. You can get it with: git clone https://github.com/OpenSIPS/opensips.git -b 1.11 [1] http://www.opensips.org/About/AvailableVersions Best regards, Liviu Chircu OpenSIPS Developerhttp://www.opensips-solutions.com On 08.03.2015 17:59, John Nash wrote: I had started testing 1.X series taken from github master branch couple of months ago (It shows version as Server:: OpenSIPS (1.12.0dev-notls (x86_64/linux)) Now I need to install it in one of the production server and before I do that I want to update to the latest version of 1.X series. Which branch should I use to use most upto date 1.X opensips (As master branch now is 2.1) ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ 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] Pike question about flood attack
As per documentation pike module can be implemented manual as well as automatic. The way I understand it manual mode will not monitor (Not even queue) packets for which pike_check_req() is not called and it gives performance advantage as we can skip this call for trusted IPs. First of all is my understanding correct? Or each request packet will be queued but we will know if a source IP exceeds threshold only when we call pike_check_req()? Second thing is what about replies, is there any way to monitor in manual mode? I really like automatic mode but only am trying to avoid it because I do not want trusted sources to be monitored. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Rtpproxy issue with serial forking
I have used opensips+rtpproxy for years for simple scenarios but now I am trying to use it with serial forking. My flow is UA---Invite ---Opensips 1 Branch ---Media Server Session Progress- 2 Branch (0n failure message)-Some Gateway --Second session progress-- My issue is SIP UA receives two session progress messages with different ports from rtpproxy. UA should receive same port in both session progress ?? isnt it?.Where should I call rtpproxy_offer and rtpproxy_answer and delete for this to achieve? John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Changing message headers (Custom header) in serial forking
I tested drouting module and it is very good but when I try to change value of one custom header (Header was added in initial Invite), I see it added twice. --Initial Invite---(Header added)Sent to dest1 ---Failure comes from dest1--- Sent to dest2 (Using use_next_gw)---Remove header if present and add new one. My guess is after I add header in failure route (after use_next_gw call) it is getting copied again from original invite. I also tried local_route but it seems its not going inside that. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Custom header parsing in failure route
I am testing a setup where opensips sending call to freeswitch and if call is rejected by freswitch a custom header X-internal-hangup. In opensips failure_route I am trying to check it using is_present_hf() function but it never reaches inside conditions. In wireshark I see this header. is_present_hf is working fine when used in main request block. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Custom header parsing in failure route
Hello Razvan, Thank you. I asked similar question few days back about status code ..sorry about that. Regards John On Thu, Jan 15, 2015 at 4:42 PM, Răzvan Crainea raz...@opensips.org wrote: Hi, John! If the call is rejected, than the X-internal-hangup header is in the reply. failure_route is ran in the request context. Therefore, you have to use the is_present_hf() function in the onreply_route. Alternatively, you can check for the header existing using the hdr pseudo-variable in the reply context (i.e if ($(replyhdr(X-internal-hangup)) != NULL)). Best regards, Răzvan Crainea OpenSIPS Solutionswww.opensips-solutions.com On 01/15/2015 12:40 PM, John Nash wrote: I am testing a setup where opensips sending call to freeswitch and if call is rejected by freswitch a custom header X-internal-hangup. In opensips failure_route I am trying to check it using is_present_hf() function but it never reaches inside conditions. In wireshark I see this header. is_present_hf is working fine when used in main request block. ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ 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] Concerns with 1.x series and new 2.1 Opensips
Ok I will test flatstore and will see how it goes if I try to insert into real DB using some other process (May be some script which will check for new records in loop) On Mon, Jan 12, 2015 at 4:06 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: John, Using exec has its own penalties - the exec itself is CPU consuming as the Operating System has to create a new process each time. Os it is not I/O, but it is CPU (system time). For the accounting part, I still recommend the flatstore as the most efficient approach. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 12.01.2015 11:55, John Nash wrote: OK..I think doing accounting in exec makes perfect sense. On Mon, Jan 12, 2015 at 2:36 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi John, Indeed, depending on the nature of the query, some answers can be cached, other not. If not, you need to be sure your DB server is as efficient as possible in answering. Accounting via flatstore file can be realtime (data is written in RT into file and you can rote them when you need). The next 2.1 is the first OpenSIPS version supporting Async I/O ops. There are many kinds of I/O ops and used in many places. It is hard to add async support for all of them from the day one. The current plan is to have support for exec module, for rest_client module and possible for some mysql queries. In the worst case, you can push your DB queries into external scripts and use the exec module with the async support. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 11.01.2015 15:48, John Nash wrote: Hello Bogdan, Thank you. Cache features are really good and I am using for Register and Invite auth but I need to run a query to find out allowed duration for a call (unfortunately caching cannot be used in that). Also Accounting I am afraid has to be real time in my case. I think i should look forward to version 2.X till the features I need are there. Any guess how long full featured development version will be out? John On Sat, Jan 10, 2015 at 1:16 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Nash, It is somewhere in the middle :). Of course the DB ops will bring some penalties to the performance, so you need to take care and tune your DB for the best performance (not to drag down opensips). With db ops is very common in OpenSIPS scripts, so you do not do anything crazy or stupid there. Of course, you should look into optimizing the DB ops you use: - DB auth - use caching at script level (see http://www.opensips.org/Documentation/Tutorials-MemoryCaching) - ACC - consider using db_flatstore to avoid writing into a real DB - dialog - if not really a must use db modes 2 or 3 ( http://www.opensips.org/html/docs/modules/1.11.x/dialog.html#id294001) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 09.01.2015 20:19, John Nash wrote: I have used opensips for load balancing and some border proxy+ NAT+rtpproxy in past and am quite happy with it. Recently I decided to add DB operations (Auth and accounting, routing and dialog into it so that heavy lifting of VOIP network can be given to opensips. I wanted to send call to PBX only when it is really needed (Like voicemail and conference etc) But in a long time I saw this article http://www.opensips.org/Documentation/TroubleShooting-FindPerfPb As per this tutorial I think if any DB operation is slow, it will hit overall performance (I mean the transactions which do not require DB can also be stuck). I know good engineers at openisps have already figured it out and working on 2.X version but looks like it will take a while so that I can give it a try (As dialog is not in current release). With 1.X series + DB auth/acc + dialog should I reconsider my approach or there are systems running successfully and I am just being paranoid? John ___ Users mailing listUsers@lists.opensips.orghttp://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] Concerns with 1.x series and new 2.1 Opensips
OK..I think doing accounting in exec makes perfect sense. On Mon, Jan 12, 2015 at 2:36 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi John, Indeed, depending on the nature of the query, some answers can be cached, other not. If not, you need to be sure your DB server is as efficient as possible in answering. Accounting via flatstore file can be realtime (data is written in RT into file and you can rote them when you need). The next 2.1 is the first OpenSIPS version supporting Async I/O ops. There are many kinds of I/O ops and used in many places. It is hard to add async support for all of them from the day one. The current plan is to have support for exec module, for rest_client module and possible for some mysql queries. In the worst case, you can push your DB queries into external scripts and use the exec module with the async support. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 11.01.2015 15:48, John Nash wrote: Hello Bogdan, Thank you. Cache features are really good and I am using for Register and Invite auth but I need to run a query to find out allowed duration for a call (unfortunately caching cannot be used in that). Also Accounting I am afraid has to be real time in my case. I think i should look forward to version 2.X till the features I need are there. Any guess how long full featured development version will be out? John On Sat, Jan 10, 2015 at 1:16 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Nash, It is somewhere in the middle :). Of course the DB ops will bring some penalties to the performance, so you need to take care and tune your DB for the best performance (not to drag down opensips). With db ops is very common in OpenSIPS scripts, so you do not do anything crazy or stupid there. Of course, you should look into optimizing the DB ops you use: - DB auth - use caching at script level (see http://www.opensips.org/Documentation/Tutorials-MemoryCaching) - ACC - consider using db_flatstore to avoid writing into a real DB - dialog - if not really a must use db modes 2 or 3 ( http://www.opensips.org/html/docs/modules/1.11.x/dialog.html#id294001) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 09.01.2015 20:19, John Nash wrote: I have used opensips for load balancing and some border proxy+ NAT+rtpproxy in past and am quite happy with it. Recently I decided to add DB operations (Auth and accounting, routing and dialog into it so that heavy lifting of VOIP network can be given to opensips. I wanted to send call to PBX only when it is really needed (Like voicemail and conference etc) But in a long time I saw this article http://www.opensips.org/Documentation/TroubleShooting-FindPerfPb As per this tutorial I think if any DB operation is slow, it will hit overall performance (I mean the transactions which do not require DB can also be stuck). I know good engineers at openisps have already figured it out and working on 2.X version but looks like it will take a while so that I can give it a try (As dialog is not in current release). With 1.X series + DB auth/acc + dialog should I reconsider my approach or there are systems running successfully and I am just being paranoid? John ___ Users mailing listUsers@lists.opensips.orghttp://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] Call sequence in serial forking
I am testing one setup where opensips drouting module sends call to Freeswitch and I encountered one situation ... UA sends Invite to opensips, opensips uses drouting module and sends Invite to Freeswitch , callee rejects the call and opensips sends ACK to freeswitch and sends second invite (from failure route). This second invite (which has same call id but different branch in via) is not treated as another transaction by freeswitch and it sends back SIP 482 Request merged response. I had the same setup tested using SEMS as SBC some times back successfully. I am not sure which side this issue should be taken care of (opensips or freeswitch) I looked in some freeswitch mail archives and in one post I can see someone suggesting that from opensips side we should increase Cseq in case of second invite. I think this can be done using script but I am not sure if i should do or not. This is the post http://lists.freeswitch.org/pipermail/freeswitch-users/2013-February/092600.html ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Concerns with 1.x series and new 2.1 Opensips
Hello Bogdan, Thank you. Cache features are really good and I am using for Register and Invite auth but I need to run a query to find out allowed duration for a call (unfortunately caching cannot be used in that). Also Accounting I am afraid has to be real time in my case. I think i should look forward to version 2.X till the features I need are there. Any guess how long full featured development version will be out? John On Sat, Jan 10, 2015 at 1:16 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Nash, It is somewhere in the middle :). Of course the DB ops will bring some penalties to the performance, so you need to take care and tune your DB for the best performance (not to drag down opensips). With db ops is very common in OpenSIPS scripts, so you do not do anything crazy or stupid there. Of course, you should look into optimizing the DB ops you use: - DB auth - use caching at script level (see http://www.opensips.org/Documentation/Tutorials-MemoryCaching) - ACC - consider using db_flatstore to avoid writing into a real DB - dialog - if not really a must use db modes 2 or 3 ( http://www.opensips.org/html/docs/modules/1.11.x/dialog.html#id294001) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 09.01.2015 20:19, John Nash wrote: I have used opensips for load balancing and some border proxy+ NAT+rtpproxy in past and am quite happy with it. Recently I decided to add DB operations (Auth and accounting, routing and dialog into it so that heavy lifting of VOIP network can be given to opensips. I wanted to send call to PBX only when it is really needed (Like voicemail and conference etc) But in a long time I saw this article http://www.opensips.org/Documentation/TroubleShooting-FindPerfPb As per this tutorial I think if any DB operation is slow, it will hit overall performance (I mean the transactions which do not require DB can also be stuck). I know good engineers at openisps have already figured it out and working on 2.X version but looks like it will take a while so that I can give it a try (As dialog is not in current release). With 1.X series + DB auth/acc + dialog should I reconsider my approach or there are systems running successfully and I am just being paranoid? John ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
