Re: [OpenSIPS-Users] What happened to auth_db ha1b?
On Mon, Sep 4, 2023 at 9:40 PM mayamatakeshi wrote: > Hi, > I'm comparing these docs: > > https://www.opensips.org/html/docs/modules/3.1.x/auth_db.html > https://www.opensips.org/html/docs/modules/3.2.x/auth_db.html > > The first one mentions ha1b but the second one doesn't mention it. > But the migration doc for 3.1 to 3.2 doesn't mention neither ha1b nor > auth_db: > https://www.opensips.org/Documentation/Migration-3-1-0-to-3-2-0 > > After update in production to 3.2, I realized a few SIP clients use > username with domain in the Authorization header and so the ha1b should be > used but how do I specify that? > > Ah. OK. I have found the explanation here: https://opensips.org/pipermail/users/2021-August/045046.html ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] What happened to auth_db ha1b?
Hi, I'm comparing these docs: https://www.opensips.org/html/docs/modules/3.1.x/auth_db.html https://www.opensips.org/html/docs/modules/3.2.x/auth_db.html The first one mentions ha1b but the second one doesn't mention it. But the migration doc for 3.1 to 3.2 doesn't mention neither ha1b nor auth_db: https://www.opensips.org/Documentation/Migration-3-1-0-to-3-2-0 After update in production to 3.2, I realized a few SIP clients use username with domain in the Authorization header and so the ha1b should be used but how do I specify that? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] compilation fails
Are you on a bash shell when you execute these commands? If not, I would try to switch to it. On Tue, Jul 4, 2023 at 2:58 AM Saint Michael wrote: > my script is > cd /usr/src/opensips-3.1/ > git reset --hard HEAD > git pull > make clean;make proper;make all > make modules > make install > and is generating errors > make[1]: Entering directory '/usr/src/opensips-3.1/modules/lua' > /bin/sh: 0: Illegal option -- > /bin/sh: 0: Illegal option -- > /bin/sh: 0: Illegal option -- > make[1]: --libs: Command not found > /bin/sh: 0: Illegal option -- > make[1]: --libs: Command not found > /bin/sh: 0: Illegal option -- > make[1]: --libs: Command not found > make[1]: Leaving directory '/usr/src/opensips-3.1/modules/lua' > > On Mon, Jul 3, 2023 at 8:43 PM Saint Michael wrote: > >> My compilation is failing by first time, version 3.1 >> is there a list of pre-requisites for compiling opensips on Ubuntu 20.04? >> >> make[1]: Entering directory '/usr/src/opensips-3.1/modules/lua' >> /bin/sh: 0: Illegal option -- >> /bin/sh: 0: Illegal option -- >> /bin/sh: 0: Illegal option -- >> make[1]: --libs: Command not found >> /bin/sh: 0: Illegal option -- >> make[1]: --libs: Command not found >> /bin/sh: 0: Illegal option -- >> make[1]: --libs: Command not found >> make[1]: Entering directory '/usr/src/opensips-3.1/modules/python' >> /bin/sh: 1: python: not found >> /bin/sh: 1: python: not found >> /bin/sh: 1: python: not found >> > ___ > 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] Remove a field from the header PAI Diversion and History info - Opensips2.4 and 3.3
On Wed, Jun 21, 2023 at 6:26 AM mayamatakeshi
wrote:
> I have not tested it but I think you could use
> {re.subst,expression}
> <https://www.kamailio.org/wiki/cookbooks/5.2.x/transformations#resubst_expression>
>
Sorry, actually, that link was for kamailio.
For opensips it is:
{re.subst,reg_exp}
<https://www.opensips.org/Documentation/Script-Tran-3-0#toc86>
> to create a variable with the value of the header with the desired field
> removed and then use append_hf to insert the variable as the modified
> header.
>
> On Tue, Jun 20, 2023 at 11:58 PM wrote:
>
>> Hello mates,
>>
>>
>>
>> Sorry for sending my message several times. I need your help
>>
>> I want to remove a field from the headers PAI, Diversion and History
>> info.
>>
>>
>>
>> I can remove the whole header by loading the *sipmsgops* and the
>> function Remove_hd
>>
>> But i did not find a way how to remove a specefic field in the header
>> PAI for example.
>>
>>
>>
>> Could you please help ?
>>
>>
>>
>> Thank you
>>
>>
>>
>> Best regards
>>
>>
>>
>> Amel
>>
>>
>> ___
>> 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] Remove a field from the header PAI Diversion and History info - Opensips2.4 and 3.3
I have not tested it but I think you could use
{re.subst,expression}
to create a variable with the value of the header with the desired field
removed and then use append_hf to insert the variable as the modified
header.
On Tue, Jun 20, 2023 at 11:58 PM wrote:
> Hello mates,
>
>
>
> Sorry for sending my message several times. I need your help
>
> I want to remove a field from the headers PAI, Diversion and History info.
>
>
>
> I can remove the whole header by loading the *sipmsgops* and the function
> Remove_hd
>
> But i did not find a way how to remove a specefic field in the header PAI
> for example.
>
>
>
> Could you please help ?
>
>
>
> Thank you
>
>
>
> Best regards
>
>
>
> Amel
>
>
> ___
> 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] WSS not sending bye
Have you confirmed a Record-Route header was added to the SIP messages
exchanged?
Anyway, I would try to enable debug log in the SIP client to see what it is
deciding to do when you press the hangup button (ideally we should see the
BYE and the address to where the request is being sent to, if it is failing
to open a WSS connection with opensips etc).
On Fri, Jun 16, 2023 at 11:22 PM Pat M wrote:
> so even after adding the record route it still does not hangup and send
> the bye
>
> Sent with Proton Mail <https://proton.me/> secure email.
>
> --- Original Message ---
> On Thursday, June 15th, 2023 at 11:02 PM, mayamatakeshi <
> mayamatake...@gmail.com> wrote:
>
>
>
> On Fri, Jun 16, 2023 at 2:05 AM Pat M wrote:
>
>> See below
>>
>> but what i can see there is no Allow, not sure if that is why?
>>
>>
>>> You are not using Record-Route.
> This is not required for SIP to work as long as the UAs are able to talk
> to each other independently of the SIP proxy after a session is established.
> But probably this is not your case.
> Please try adding this to your route{}:
>
> if (is_method("INVITE"))
> record_route();
>
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] WSS not sending bye
On Fri, Jun 16, 2023 at 2:05 AM Pat M wrote:
> See below
>
> but what i can see there is no Allow, not sure if that is why?
>
>
>> You are not using Record-Route.
This is not required for SIP to work as long as the UAs are able to talk to
each other independently of the SIP proxy after a session is established.
But probably this is not your case.
Please try adding this to your route{}:
if (is_method("INVITE"))
record_route();
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] WSS not sending bye
As Răzvan said,
please post the SIP logs/capture (the SIP messages).
Probably there is some issue with Record-Route headers.
On Thu, Jun 15, 2023 at 12:49 AM Pat M via Users
wrote:
> anyone have any advice what i can check?
> i am desperate now
>
>
>
>
> Sent with Proton Mail secure email.
>
> --- Original Message ---
> On Wednesday, June 14th, 2023 at 10:05 AM, Pat M via Users <
> users@lists.opensips.org> wrote:
>
>
> > Hi Razvan
> >
> > I do not see anything in the logs of the client but here is the invite
> portion of the opensips
> > outbound calls have no problem it is only calls the webrtc client
> receives
> >
> >
> > route{
> > if(!has_totag() && is_method("INVITE")) {
> > if ($rU =~ '^[0-9]{11,}$') {
> > create_dialog();
> > topology_hiding("UC");
> >
> >
> >
> > $var(rtpengine_params) = $var(direction) + " replace-origin
> replace-session-connection";
> >
> > if ( has_body("application/sdp") )
> > rtpengine_manage("$var(rtpengine_params)");
> >
> > $du = "sip:" + :5060;
> > $ru = "sip:" + $rU + "@" + :5060;
> >
> > route(RELAY);
> > } else {
> > xlog("Not Outbound Call");
> > }
> > xlog("($$ct=$tU)\n");
> >
> > if (!mid_registrar_lookup("location")) {
> > t_reply(404, "Not Found");
> > exit;
> > }
> > create_dialog();
> > topology_hiding("U");
> > $var(rtpengine_params) = "trust-address replace-origin
> replace-session-connection rtcp-mux-offer ICE=force transcode-PCMU
> transcode-G722 SDES-off UDP/TLS/RTP/SAVPF";
> > xlog("USING RTPENGINE!!!");
> > if ( has_body("application/sdp") )
> > rtpengine_manage("$var(rtpengine_params)");
> > t_relay();
> > exit;
> >
> > }
> >
> >
> >
> >
> >
> >
> > Sent with Proton Mail secure email.
> >
> >
> > --- Original Message ---
> > On Wednesday, June 14th, 2023 at 8:53 AM, Răzvan Crainea
> raz...@opensips.org wrote:
> >
> >
> >
> > > Hi, Pat!
> > >
> > > You might be missing a Record-Route or something. Please post the SIP
> > > logs from your WebRTC client if you need further help.
> > >
> > > Best regards,
> > >
> > > Răzvan Crainea
> > > OpenSIPS Core Developer
> > > http://www.opensips-solutions.com
> > >
> > > On 6/13/23 19:56, Pat M via Users wrote:
> > >
> > > > Hi everyone
> > > >
> > > > Please help!!
> > > >
> > > > I have a mobile webrtc client that i am trying to use with opensips,
> it
> > > > all works fine except when you try hangup from the client it does
> > > > nothing, does not send any bye to opensips and therefore opensips
> does
> > > > not send a bye upstream
> > > > i am using mid_registrar
> > > >
> > > > but if i register it direct to an asterisk server it works fine
> > > >
> > > > my wss config is very normal including the tls section
> > > >
> > > > loadmodule "proto_wss.so"
> > > > modparam("proto_wss", "require_origin", yes)
> > > > modparam("proto_wss", "wss_max_msg_chunks", 16)
> > > > modparam("proto_wss", "wss_handshake_timeout", 300)
> > > >
> > > > modparam("tls_mgm", "ca_list", "[sip]/etc/letsencrypt/fullchain.pem")
> > > > modparam("tls_mgm", "certificate", "[sip]/etc/opensips/tls/cert.pem")
> > > > modparam("tls_mgm", "private_key", "[sip]/etc/opensips/tls/ckey.pem")
> > > > modparam("tls_mgm", "require_cert", "[sip]0")
> > > > modparam("tls_mgm", "verify_cert", "[sip]0")
> > > > modparam("tls_mgm", "tls_library", "wolfssl")
> > > >
> > > > modparam("tls_mgm", "client_domain", "sip1")
> > > > modparam("tls_mgm", "ca_list",
> "[sip1]/etc/letsencrypt/fullchain.pem")
> > > > modparam("tls_mgm", "certificate",
> "[sip1]/etc/opensips/tls/sip1.pem")
> > > > modparam("tls_mgm", "private_key",
> "[sip1]/etc/opensips/tls/sip1key.pem")
> > > > modparam("tls_mgm", "require_cert", "[sip1]0")
> > > > modparam("tls_mgm", "verify_cert", "[sip1]0")
> > > > modparam("tls_mgm", "match_ip_address", "[sip1]*")
> > > >
> > > > loadmodule "mid_registrar.so"
> > > > modparam("mid_registrar", "mode", 2) /* 1 = mirror / 1 = ct / 2 =
> AoR */
> > > > modparam("mid_registrar", "outgoing_expires", 180)
> > > > modparam("mid_registrar", "min_expires", 60)
> > > > modparam("mid_registrar", "max_expires", 180)
> > > > modparam("mid_registrar", "max_contacts", 16)
> > > > modparam("mid_registrar", "received_avp", "$avp(received)")
> > > > modparam("mid_registrar", "pn_pnsreg_interval", 140)
> > > >
> > > > mid_registrar_save("location", "p0v");
> > > >
> > > > Thanks Pat
> > > > Sent with Proton Mail https://proton.me/ secure email.
> > > >
> > > > ___
> > > > 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] How to CANCEL a Dialog from script
Maybe you could use t_cancel_branch():
onreply_route[XXX] {
...
if ((t_check_status("180") && (SOME_HEADER_HAS_SOME_VALUE)) {
t_cancel_branch("a");
}
...
}
https://opensips.org/html/docs/modules/3.2.x/tm#func_t_cancel_branch
On Mon, Nov 7, 2022 at 6:45 PM Alain Bieuzent
wrote:
> Hi all,
>
>
>
> I need to end a call not yet established from my script; how can I do that?
>
>
>
> example :
>
> I receive an INVITE followed by a 180 Ringing
>
> I read a header in the 180 Ringing and depending on the value of the
> header, I want to cancel this call.
>
>
>
> Kinda like like a dlg_end_dlg but from my script.
>
>
>
> Alain.
> ___
> 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] Orderly shutdown
It can be done with:
https://opensips.org/docs/modules/3.1.x/gflags.html
On Thu, Oct 27, 2022 at 1:18 PM Saint Michael wrote:
> Dear Friends
> I successfully wrote a script that terminates all open calls in an
> orderly fashion.
> The question is, before I execute my script, is there any similar
> command (like opensips-cli -x mi dlg_end_dlg "${callid}") that would
> stop taking new calls? Like "reject any call attempts with 503 or
> 404", because, in a high-pressure environment, 2000 new calls may
> arrive before I close all calls and shut down Opensips.
> I use version:
> opensips -V
> version: opensips 3.1.12 (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_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, sigio_rt, select.
> git revision: 15ef14a89
> main.c compiled on 03:16:12 Oct 25 2022 with gcc 9
>
> On Wed, Oct 26, 2022 at 10:49 AM Bogdan-Andrei Iancu
> wrote:
> >
> > Hi,
> >
> > Sending a sig TERM to the opensips processes will trigger a controlled
> > shutdown of opensips (meaning with flushing data to DB, if needed,
> > cleanup, etc). Nevertheless, the opensips shutdown does not ends the
> > call, as the calls may be continued after restarting opensips.
> >
> > Still, if you want to terminate all ongoing calls before shutting down,
> > better use dialog module and check these MI cmds:
> > https://opensips.org/html/docs/modules/3.2.x/dialog.html#mi_dlg_end_dlg
> >
> https://opensips.org/html/docs/modules/3.2.x/dialog.html#mi_profile_end_dlgs
> >
> > Regards,
> >
> > Bogdan-Andrei Iancu
> >
> > OpenSIPS Founder and Developer
> >https://www.opensips-solutions.com
> > OpenSIPS Bootcamp 5-16 Dec 2022, online
> >https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
> >
> > On 10/25/22 2:15 PM, Saint Michael wrote:
> > > Dear friends
> > > is there a command I can send to Opensips 3.1, so all calls are shut
> > > down and BYEs are sent to both the caller and the callee? Then of
> > > course, opensips closes down.
> > >
> > > ___
> > > 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] early dialog termination
Hi,
I've been following this discussion as I have a similar use case.
I have created a PR offering a new function t_reply_by_callid for the
module tm to simplify this:
https://github.com/OpenSIPS/opensips/pull/2937
Regards,
Takeshi
On Thu, Oct 20, 2022 at 4:04 PM Bogdan-Andrei Iancu
wrote:
> Ivan,
>
> Actually a simpler approach will be to use t_wait_for_new_branches()
> instead of that t_write function, it should do the same trick (postponing
> the deletion of the transaction), but without any side effects.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Bootcamp 5-16 Dec 2022, online
> https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 10/19/22 10:21 AM, Ryzhik Ivan wrote:
>
> Sorry, I mean no sleep, i mean async( sleep($var(wait_time)), after_sleep
> );
> Regards, Ivan.
>
> вт, 18 окт. 2022 г. в 14:42, Bogdan-Andrei Iancu :
>
>> Hi,
>>
>> yes, call it before ending the REQUEST route. I'm 100% the transaction is
>> not deleted before the end of the route. And try to use the unix sock
>> flavor for the function, not the fifo one.
>>
>> DO NOT use the sleep, you will block your whole opensips.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>> https://www.opensips-solutions.com
>> OpenSIPS Bootcamp 5-16 Dec 2022, online
>> https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>>
>> On 10/17/22 11:56 AM, Ryzhik Ivan wrote:
>>
>> Hi, did you mean that i must call t_write_req once before REQUEST_ROUTE
>> is finished? In this case the transaction was removed.
>> "even if you do not have to actually write anything to outer world -
>> just fake it." - i must use fifo and i must read data from it, in else
>> we got:
>> ERROR:tm:write_to_fifo: nobody listening on [/tmp/moh.fifo] fifo for
>> reading!
>> ERROR:tm:t_write_req: write_to_fifo failed
>> And last question is may I use sleep(20) at the end of route to keep
>> transaction? or can i use modparam("tm", "wt_timer", 20)?
>> Regards, Ivan
>>
>> пн, 17 окт. 2022 г. в 09:38, Bogdan-Andrei Iancu :
>>
>>> Hi Ryzhik,
>>>
>>> Right, the transaction must be forced to stay until you are done with a
>>> final reply. Unfortunately there is no clear way to do this from script
>>> (this may be subject of further small improvements), but you can try taking
>>> advantage of the `t_write_req` [1] for this purpose, even if you do not
>>> have to actually write anything to outer world - just fake it.
>>>
>>>
>>> [1]
>>> https://opensips.org/html/docs/modules/3.2.x/tm.html#func_t_write_req
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>> https://www.opensips-solutions.com
>>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>> https://www.opensips.org/events/Summit-2022Athens/
>>>
>>> On 10/13/22 2:45 PM, Ryzhik Ivan wrote:
>>>
>>> Hi.
>>> One more question.
>>> Everything works fine except the transaction was deleted after 15
>>> sec after the initial route was finished.
>>> on INVITE i last do t_reply_with_body(183, "Session progress", ...) and
>>> than exit;
>>>
>>> on end route log :
>>>
>>> 2022-10-13T10:58:01.994598+00:00 DBG:tm:_reply_light: reply sent out.
>>> buf=0x7f558a087d98: SIP/2.0 1..., shmem=0x7f5549797470: SIP/2.0 1
>>> 2022-10-13T10:58:01.994676+00:00 DBG:tm:_reply_light: finished
>>>
>>> 2022-10-13T10:58:01.995835+00:00 DBG:tm:do_t_cleanup: transaction
>>> 0x7f5549793b18 already updated! Skipping update!
>>> 2022-10-13T10:58:01.996020+00:00 DBG:tm:cleanup_uac_timers: RETR/FR
>>> timers reset
>>> 2022-10-13T10:58:01.996202+00:00 *DBG:tm:insert_timer_unsafe: [2]:
>>> 0x7f5549793b98 (12)*
>>> 2022-10-13T10:58:01.996317+00:00 * DBG:tm:t_unref: UNREF_UNSAFE:
>>> [0x7f5549793b18] after is 0*
>>> 2022-10-13T10:58:01.996488+00:00 DBG:core:destroy_avp_list: destroying
>>> list (nil)
>>> 2022-10-13T10:58:01.996673+00:00 DBG:core:receive_msg: cleaning up
>>>
>>> 2022-10-13T10:58:07.651091+00:00* DBG:tm:timer_routine: timer
>>> routine:2,tl=0x7f5549793b98 next=(nil), timeout=12*
>>> 2022-10-13T10:58:07.651332+00:00 DBG:tm:wait_handler: removing
>>> 0x7f5549793b18 from table
>>> 2022-10-13T10:58:07.651425+00:00 DBG:tm:delete_ce*ll: delete
>>> transaction 0x7f5549793b18*
>>> 2022-10-13T10:58:07.651513+00:00 DBG:tm:wait_handler: done
>>>
>>> Can you tell me how I can i fix this? Transaction marked safe for
>>> deletion...
>>> Regards, Ivan
>>>
>>> ср, 12 окт. 2022 г. в 13:11, Bogdan-Andrei Iancu :
>>>
Perfect !!!
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
https://www.opensips.org/events/Summit-2022Athens/
On 10/12/22 1:09 PM, Ryzhik Ivan wrote:
I found a solution. hex strings are reversed).
Thank you very much!
ср, 12 окт. 2022 г. в 12:59, Ryzhik Ivan :
> and one more research: $T_id returns hex encoded label.hashid
> but
Re: [OpenSIPS-Users] Creating branches inside a while loop
Hi Bogdan,
yes. It works.
I am using it like this:
*$var(idx) = 0;$var(aor) =
$(var(aors){s.select,$var(idx),,});while($var(aor)
!= null) {if($var(idx) > 0) {
append_branch();}
seturi("sip:$var(aor)@$fd");$var(idx) = $var(idx) + 1;
$var(aor) = $(var(aors){s.select,$var(idx),!});}
if (lookup("location", "r")) {
t_on_branch("USERAGENT_BRANCH");route(RELAY_REQUEST);
} else {sl_send_reply(480, "subscriber not online
retcode=$retcode");}*
Thanks and regards,
Takeshi
On Tue, Oct 11, 2022 at 6:32 PM Bogdan-Andrei Iancu
wrote:
> Hi, and it works now, right?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
> https://www.opensips.org/events/Summit-2022Athens/
>
> On 10/11/22 10:29 AM, mayamatakeshi wrote:
>
> Hi Bogdan,
> indeed, I think I removed the append_branch() line by mistake when
> removing some debug lines I added to check function return code etc.
> Thanks,
> Takeshi
>
> On Mon, Oct 10, 2022 at 9:39 PM Bogdan-Andrei Iancu
> wrote:
>
>> Hi Takeshi,
>>
>> Aren't you missing the "seturi()" + "append_branch()" in the loop ??
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>> https://www.opensips-solutions.com
>> OpenSIPS Summit 27-30 Sept 2022, Athens
>> https://www.opensips.org/events/Summit-2022Athens/
>>
>> On 9/28/22 4:26 PM, mayamatakeshi wrote:
>>
>>
>> On Wed, Sep 28, 2022 at 2:21 PM mayamatakeshi
>> wrote:
>>
>>> Hi,
>>> I'm testing latest commit b243666098be44226ade6a7df2b62851efcb5de8 of
>>> opensips-3.2.
>>>
>>> I tested adding branches to an INVITE for a fixed size list of AORs this
>>> way:
>>>
>>> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
>>> sip:us...@test1.com";
>>>
>>> seturi($(var(aors){s.select,0,,}));
>>>
>>> append_branch();
>>> seturi($(var(aors){s.select,1,,}));
>>>
>>> append_branch();
>>> seturi($(var(aors){s.select,2,,}));
>>>
>>> lookup("location", "r")
>>>
>>> The above works fine and all 3 destinations resolved by AOR lookup are
>>> called (max of contact per AOR).
>>>
>>> However, in case of a a list of unknown size, I tried to use a while
>>> loop like this:
>>> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
>>> sip:us...@test1.com";
>>>
>>> $var(idx) = 0;
>>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>>
>>> while($var(aor) != null) {
>>> seturi($var(aor));
>>>
>>> $var(idx) = $var(idx) + 1;
>>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>> }
>>>
>>> lookup("location", "r")
>>>
>>> But with the above, only the last destination (lookup of us...@test1.com)
>>> is called.
>>> I confirmed this is not related to the lookup function because I tried
>>> with fixed destinations like this:
>>>
>>> $var(aors) = "sip:user1@10.0.0.1:5072,
>>> sip:user2@10.0.0.1:5074,sip:user3@10.0.0.1:5076";
>>>
>>> $var(idx) = 0;
>>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>>
>>> while($var(aor) != null) {
>>> seturi($var(aor));
>>>
>>> $var(idx) = $var(idx) + 1;
>>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>> }
>>>
>>> and the same problem happens: only the last destination
>>> sip:user3@10.0.0.1:5076 is called.
>>>
>>> So, is there a way to append a non-fixed number of branches to an INVITE?
>>>
>>> Regards,
>>> Takeshi
>>>
>>
>> Sorry, I think I did something wrong.
>> I was able to make append_branch to work inside a while loop.
>> So there is no problem.
>> Regards,
>> Takeshi
>>
>>
>> ___
>> 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] Creating branches inside a while loop
Hi Bogdan,
indeed, I think I removed the append_branch() line by mistake when removing
some debug lines I added to check function return code etc.
Thanks,
Takeshi
On Mon, Oct 10, 2022 at 9:39 PM Bogdan-Andrei Iancu
wrote:
> Hi Takeshi,
>
> Aren't you missing the "seturi()" + "append_branch()" in the loop ??
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
> https://www.opensips.org/events/Summit-2022Athens/
>
> On 9/28/22 4:26 PM, mayamatakeshi wrote:
>
>
> On Wed, Sep 28, 2022 at 2:21 PM mayamatakeshi
> wrote:
>
>> Hi,
>> I'm testing latest commit b243666098be44226ade6a7df2b62851efcb5de8 of
>> opensips-3.2.
>>
>> I tested adding branches to an INVITE for a fixed size list of AORs this
>> way:
>>
>> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
>> sip:us...@test1.com";
>>
>> seturi($(var(aors){s.select,0,,}));
>>
>> append_branch();
>> seturi($(var(aors){s.select,1,,}));
>>
>> append_branch();
>> seturi($(var(aors){s.select,2,,}));
>>
>> lookup("location", "r")
>>
>> The above works fine and all 3 destinations resolved by AOR lookup are
>> called (max of contact per AOR).
>>
>> However, in case of a a list of unknown size, I tried to use a while loop
>> like this:
>> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
>> sip:us...@test1.com";
>>
>> $var(idx) = 0;
>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>
>> while($var(aor) != null) {
>> seturi($var(aor));
>>
>> $var(idx) = $var(idx) + 1;
>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>> }
>>
>> lookup("location", "r")
>>
>> But with the above, only the last destination (lookup of us...@test1.com)
>> is called.
>> I confirmed this is not related to the lookup function because I tried
>> with fixed destinations like this:
>>
>> $var(aors) = "sip:user1@10.0.0.1:5072,sip:user2@10.0.0.1:5074
>> ,sip:user3@10.0.0.1:5076";
>>
>> $var(idx) = 0;
>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>
>> while($var(aor) != null) {
>> seturi($var(aor));
>>
>> $var(idx) = $var(idx) + 1;
>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>> }
>>
>> and the same problem happens: only the last destination
>> sip:user3@10.0.0.1:5076 is called.
>>
>> So, is there a way to append a non-fixed number of branches to an INVITE?
>>
>> Regards,
>> Takeshi
>>
>
> Sorry, I think I did something wrong.
> I was able to make append_branch to work inside a while loop.
> So there is no problem.
> Regards,
> Takeshi
>
>
> ___
> 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] Creating branches inside a while loop
On Wed, Sep 28, 2022 at 2:21 PM mayamatakeshi
wrote:
> Hi,
> I'm testing latest commit b243666098be44226ade6a7df2b62851efcb5de8 of
> opensips-3.2.
>
> I tested adding branches to an INVITE for a fixed size list of AORs this
> way:
>
> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
> sip:us...@test1.com";
>
> seturi($(var(aors){s.select,0,,}));
>
> append_branch();
> seturi($(var(aors){s.select,1,,}));
>
> append_branch();
> seturi($(var(aors){s.select,2,,}));
>
> lookup("location", "r")
>
> The above works fine and all 3 destinations resolved by AOR lookup are
> called (max of contact per AOR).
>
> However, in case of a a list of unknown size, I tried to use a while loop
> like this:
> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
> sip:us...@test1.com";
>
> $var(idx) = 0;
> $var(aor) = $(var(aors){s.select,$var(idx),,});
>
> while($var(aor) != null) {
> seturi($var(aor));
>
> $var(idx) = $var(idx) + 1;
> $var(aor) = $(var(aors){s.select,$var(idx),,});
> }
>
> lookup("location", "r")
>
> But with the above, only the last destination (lookup of us...@test1.com)
> is called.
> I confirmed this is not related to the lookup function because I tried
> with fixed destinations like this:
>
> $var(aors) = "sip:user1@10.0.0.1:5072,sip:user2@10.0.0.1:5074,
> sip:user3@10.0.0.1:5076";
>
> $var(idx) = 0;
> $var(aor) = $(var(aors){s.select,$var(idx),,});
>
> while($var(aor) != null) {
> seturi($var(aor));
>
> $var(idx) = $var(idx) + 1;
> $var(aor) = $(var(aors){s.select,$var(idx),,});
> }
>
> and the same problem happens: only the last destination
> sip:user3@10.0.0.1:5076 is called.
>
> So, is there a way to append a non-fixed number of branches to an INVITE?
>
> Regards,
> Takeshi
>
Sorry, I think I did something wrong.
I was able to make append_branch to work inside a while loop.
So there is no problem.
Regards,
Takeshi
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Route header in CANCEL requests
Hi, i have opensips behind another SIP proxy. When I need to t_relay an INVITE from an uac to the proxy, my opensips.cfg uses append_hf to add a Route header with a username required by the proxy (some sort of validation). This works fine. However, if the uac cancels the INVITE, the transaction-controlled CANCEL generated by opensips goes out without this Route header. I was expecting opensips to add this Route header because opensips is also used as a registrar and the proxy in front of it adds Path header to the REGISTER requests and then when I relay an INVITE to the registered entity (after doing lookup() ), the INVITE goes out with the contents of the Path header in a Route header and the same Route header is added by opensips to the CANCEL it generates. So, is there a way to force opensips to add this header to CANCEL? (I cannot change the proxy in front of opensips. I need to conform with it). I am using opensips 1.11 commit 719f298d5ad3300b790f64cfed40e797eeb23225. Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Route header in CANCEL requests
Hello,
i have found the solution in the docs:
append_hf("Route: <sip:x...@test.com>\r\n");
t_newtran();
t_add_hdrs("Route: <sip:z...@test.com>\r\n");
Regards,
Takeshi
On Sat, Oct 10, 2015 at 9:03 AM, mayamatakeshi <mayamatake...@gmail.com>
wrote:
> Hi, i have opensips behind another SIP proxy.
> When I need to t_relay an INVITE from an uac to the proxy, my opensips.cfg
> uses append_hf to add a Route header with a username required by the proxy
> (some sort of validation).
> This works fine. However, if the uac cancels the INVITE, the
> transaction-controlled CANCEL generated by opensips goes out without this
> Route header.
> I was expecting opensips to add this Route header because opensips is also
> used as a registrar and the proxy in front of it adds Path header to the
> REGISTER requests and then when I relay an INVITE to the registered entity
> (after doing lookup() ), the INVITE goes out with the contents of the Path
> header in a Route header and the same Route header is added by opensips to
> the CANCEL it generates.
>
> So, is there a way to force opensips to add this header to CANCEL? (I
> cannot change the proxy in front of opensips. I need to conform with it).
> I am using opensips 1.11 commit 719f298d5ad3300b790f64cfed40e797eeb23225.
> Regards,
> Takeshi
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Route header in CANCEL requests
For completeness, it will be also necessary to call t_reply to send '100
Trying' as we are using t_newtran():
append_hf("Route: <sip:x...@test.com>\r\n");
t_newtran();
t_reply("100", "Trying");
t_add_hdrs("Route: <sip:z...@test.com>\r\n");
On Sat, Oct 10, 2015 at 10:27 AM, mayamatakeshi <mayamatake...@gmail.com>
wrote:
> Hello,
> i have found the solution in the docs:
>
> append_hf("Route: <sip:x...@test.com>\r\n");
> t_newtran();
> t_add_hdrs("Route: <sip:z...@test.com>\r\n");
>
> Regards,
> Takeshi
>
> On Sat, Oct 10, 2015 at 9:03 AM, mayamatakeshi <mayamatake...@gmail.com>
> wrote:
>
>> Hi, i have opensips behind another SIP proxy.
>> When I need to t_relay an INVITE from an uac to the proxy, my
>> opensips.cfg uses append_hf to add a Route header with a username required
>> by the proxy (some sort of validation).
>> This works fine. However, if the uac cancels the INVITE, the
>> transaction-controlled CANCEL generated by opensips goes out without this
>> Route header.
>> I was expecting opensips to add this Route header because opensips is
>> also used as a registrar and the proxy in front of it adds Path header to
>> the REGISTER requests and then when I relay an INVITE to the registered
>> entity (after doing lookup() ), the INVITE goes out with the contents of
>> the Path header in a Route header and the same Route header is added by
>> opensips to the CANCEL it generates.
>>
>> So, is there a way to force opensips to add this header to CANCEL? (I
>> cannot change the proxy in front of opensips. I need to conform with it).
>> I am using opensips 1.11 commit 719f298d5ad3300b790f64cfed40e797eeb23225.
>> Regards,
>> Takeshi
>>
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No acc records after opensips restart
On Mon, May 5, 2014 at 11:43 PM, mayamatakeshi mayamatake...@gmail.comwrote: On Mon, May 5, 2014 at 10:16 PM, Răzvan Crainea raz...@opensips.orgwrote: I think the first patch was corrupted while I pasted it. Please try to use this one[1]. [1] http://pastebin.com/DvRCwT20 Hi Razvan, Actually, it seems that message from patch was just a warning and can be ignored ( http://unix.stackexchange.com/questions/1395/what-does-patch-unexpectedly-ends-in-middle-of-line-mean ). So i checked the code and the problem was a typo. Instead of : LM_DBG(Registered terminated callback with flags %x\n, reg-flags); it should have been: LM_DBG(Registered terminated callback with flags %x\n, req-flags); So I corrected it and rebuilt opensips and tested it. Here are the logs: http://pastebin.com/Ne3vUdzG Razvan, sorry, I checked the logs and the lines you added are not being printed. It seems I am loading the modules from the wrong path. I will correct it and get the logs again. Please wait. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No acc records after opensips restart
Hi Razvan, On Wed, May 7, 2014 at 5:10 PM, Răzvan Crainea raz...@opensips.org wrote: Hi, Takeshi! I am not sure you patch was successfully installed, since I don't see the logs I put in the patch. Sorry. I was loading the modules from the wrong path so it was using unpatched build. Here are the logs: http://pastebin.com/5ubakRkP (but I only see the Registered terminated callback with flags line in them). Is there any chance you could give me access to a development machine where this is happening? If it gets difficult for you to debug this from the logs only, i will try to prepare an environment at home because the company i work for doesn't allow me to give this access. Regards, Takeshi Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 05/05/2014 05:43 PM, mayamatakeshi wrote: On Mon, May 5, 2014 at 10:16 PM, Răzvan Crainea raz...@opensips.org mailto:raz...@opensips.org wrote: I think the first patch was corrupted while I pasted it. Please try to use this one[1]. [1] http://pastebin.com/DvRCwT20 Hi Razvan, Actually, it seems that message from patch was just a warning and can be ignored (http://unix.stackexchange.com/questions/1395/what-does- patch-unexpectedly-ends-in-middle-of-line-mean). So i checked the code and the problem was a typo. Instead of : LM_DBG(Registered terminated callback with flags %x\n, reg-flags); it should have been: LM_DBG(Registered terminated callback with flags %x\n, req-flags); So I corrected it and rebuilt opensips and tested it. Here are the logs: http://pastebin.com/Ne3vUdzG Regards, Takeshi ___ 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] No acc records after opensips restart
Razvan, sorry again. I did a dumb mistake and used opensipsctl fifo debug 9 instead of enabling the debug in the opensips.cfg file. So when opensips restarted, debug was disabled. Here are the logs: http://pastebin.com/mtG6n68E Regards, Takeshi On Thu, May 8, 2014 at 3:06 AM, Răzvan Crainea raz...@opensips.org wrote: Hi, Takeshi! I don't see any BYE or restart in your logs. Can you please send the logs starting from the INVITE, then restart and BYE? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 05/07/2014 11:58 AM, mayamatakeshi wrote: Hi Razvan, On Wed, May 7, 2014 at 5:10 PM, Răzvan Crainea raz...@opensips.org mailto:raz...@opensips.org wrote: Hi, Takeshi! I am not sure you patch was successfully installed, since I don't see the logs I put in the patch. Sorry. I was loading the modules from the wrong path so it was using unpatched build. Here are the logs: http://pastebin.com/5ubakRkP (but I only see the Registered terminated callback with flags line in them). Is there any chance you could give me access to a development machine where this is happening? If it gets difficult for you to debug this from the logs only, i will try to prepare an environment at home because the company i work for doesn't allow me to give this access. Regards, Takeshi Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.__com http://www.opensips- solutions.com On 05/05/2014 05:43 PM, mayamatakeshi wrote: On Mon, May 5, 2014 at 10:16 PM, Răzvan Crainea raz...@opensips.org mailto:raz...@opensips.org mailto:raz...@opensips.org mailto:raz...@opensips.org wrote: I think the first patch was corrupted while I pasted it. Please try to use this one[1]. [1] http://pastebin.com/DvRCwT20 Hi Razvan, Actually, it seems that message from patch was just a warning and can be ignored (http://unix.stackexchange.__com/questions/1395/what-does-_ _patch-unexpectedly-ends-in-__middle-of-line-mean http://unix.stackexchange.com/questions/1395/what-does- patch-unexpectedly-ends-in-middle-of-line-mean). So i checked the code and the problem was a typo. Instead of : LM_DBG(Registered terminated callback with flags %x\n, reg-flags); it should have been: LM_DBG(Registered terminated callback with flags %x\n, req-flags); So I corrected it and rebuilt opensips and tested it. Here are the logs: http://pastebin.com/Ne3vUdzG Regards, Takeshi _ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-__bin/mailman/listinfo/users http://lists.opensips.org/cgi-bin/mailman/listinfo/users _ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-__bin/mailman/listinfo/users 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] No acc records after opensips restart
Hi Razvan, I will do it. But how should I apply the patch? I tried using patch command but I got an error: root@LAB002177-STR2-SERVER:opensips-1.11# patch -p0 razvan_patch.txt (Stripping trailing CRs from patch.) patching file modules/acc/acc.c (Stripping trailing CRs from patch.) patching file modules/acc/acc_logic.c (Stripping trailing CRs from patch.) patching file modules/dialog/dlg_db_handler.c patch unexpectedly ends in middle of line Hunk #2 succeeded at 376 with fuzz 1. Anyway, I tried to compile this but I got an error: make[1]: Entering directory `/usr/local/src/git/opensips-1.11/modules/acc' Compiling acc.c Compiling acc_logic.c acc_logic.c: In function ‘acc_onreply’: acc_logic.c:597:3: error: ‘reg’ undeclared (first use in this function) acc_logic.c:597:3: note: each undeclared identifier is reported only once for each function it appears in make[1]: *** [acc_logic.o] Error 1 make[1]: *** Waiting for unfinished jobs make[1]: Leaving directory `/usr/local/src/git/opensips-1.11/modules/acc' make: *** [modules] Error 2 I also inspected the patch with command 'git apply' but i go this: root@LAB002177-STR2-SERVER:opensips-1.11# git apply --stat razvan_patch.txt fatal: corrupt patch at line 72 I don't know much about patch file formats. Please instruct me on how to apply the patch. Regards, Takeshi On Mon, May 5, 2014 at 7:06 PM, Răzvan Crainea raz...@opensips.org wrote: Hi, Takeshi! Can you please apply this patch [1] and run again in debugging mode? Thanks, [1] http://pastebin.com/i6uvbREf Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 05/01/2014 03:26 PM, mayamatakeshi wrote: Hello Razvan, On Thu, May 1, 2014 at 4:11 PM, Răzvan Crainea raz...@opensips.org mailto:raz...@opensips.org wrote: Hi, Takeshi! You are right, CDR_FLAG should only be set on the initial INVITE. DB_FLAG should not be set for BYE, because otherwise you will only account that method, not the CDR (no duration, setuptime, etc.). Have you tried with other Accounting backends (log, events)? Yes. I use both db and log backends and no CDRs are written to opensips.log in 1.11 after an opensips restart. Can you run a test in debugging mode, paste the logs on pastebin[1] and send them over (privately if you want)? [1] http://pastebin.com/ Here it is: http://pastebin.com/hcdybZL3 Thanks. Regards, Takeshi Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.__com http://www.opensips- solutions.com On 05/01/2014 09:23 AM, mayamatakeshi wrote: Razvan, no, I was only setting the CDR_FLAG on the initial INVITE. This works with 1.10. Then with 1.11, I tested setting the CDR_FLAG when processing BYE requests but it didn't work. Then I set DB_FLAG for BYE and after that, it worked. So this behavior changed. Thanks a lot. Regards, Takeshi. On Wed, Apr 30, 2014 at 11:05 PM, Răzvan Crainea raz...@opensips.org mailto:raz...@opensips.org mailto:raz...@opensips.org mailto:raz...@opensips.org wrote: Hi, Takeshi! I just tested and there are no issues on my setup. Are you setting the CDR_FLAG for all the requests? Have you also tried with different db modes? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com http://www.opensips-__solutions.com http://www.opensips-solutions.com On 04/23/2014 03:05 PM, mayamatakeshi wrote: Hello, I am trying to upgrade from opensips 1.10 (commit bfd86ab25554082053167a82655e669ca1c0ea7c) to 1.11 (latest 313f41ad46921d543980fa9fc71fb0715d385085). Almost everything works. However, in my opensips.cfg I set dialog db_mode to REALTIME. And I have acc in CDR mode. Then after making some calls and confirming the dialogs were inserted in the opensips.dialog table, I kill and restart opensips. Then I terminate the calls. In 1.10, this causes proper generation of records in the acc table. However, in 1.11, no records are written to acc table. I checked the release notes for 1.11 but could not find any indication that this behavior would change: http://opensips.org/pub/opensips/1.11.0/src/ChangeLog http://opensips.org/pub/__opensips/1.11.0/src/ChangeLog http://opensips.org/pub/__opensips/1.11.0/src/ChangeLog http
Re: [OpenSIPS-Users] No acc records after opensips restart
On Mon, May 5, 2014 at 10:16 PM, Răzvan Crainea raz...@opensips.org wrote: I think the first patch was corrupted while I pasted it. Please try to use this one[1]. [1] http://pastebin.com/DvRCwT20 Hi Razvan, Actually, it seems that message from patch was just a warning and can be ignored ( http://unix.stackexchange.com/questions/1395/what-does-patch-unexpectedly-ends-in-middle-of-line-mean ). So i checked the code and the problem was a typo. Instead of : LM_DBG(Registered terminated callback with flags %x\n, reg-flags); it should have been: LM_DBG(Registered terminated callback with flags %x\n, req-flags); So I corrected it and rebuilt opensips and tested it. Here are the logs: http://pastebin.com/Ne3vUdzG Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No acc records after opensips restart
Hello Razvan, On Thu, May 1, 2014 at 4:11 PM, Răzvan Crainea raz...@opensips.org wrote: Hi, Takeshi! You are right, CDR_FLAG should only be set on the initial INVITE. DB_FLAG should not be set for BYE, because otherwise you will only account that method, not the CDR (no duration, setuptime, etc.). Have you tried with other Accounting backends (log, events)? Yes. I use both db and log backends and no CDRs are written to opensips.log in 1.11 after an opensips restart. Can you run a test in debugging mode, paste the logs on pastebin[1] and send them over (privately if you want)? [1] http://pastebin.com/ Here it is: http://pastebin.com/hcdybZL3 Thanks. Regards, Takeshi Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 05/01/2014 09:23 AM, mayamatakeshi wrote: Razvan, no, I was only setting the CDR_FLAG on the initial INVITE. This works with 1.10. Then with 1.11, I tested setting the CDR_FLAG when processing BYE requests but it didn't work. Then I set DB_FLAG for BYE and after that, it worked. So this behavior changed. Thanks a lot. Regards, Takeshi. On Wed, Apr 30, 2014 at 11:05 PM, Răzvan Crainea raz...@opensips.org mailto:raz...@opensips.org wrote: Hi, Takeshi! I just tested and there are no issues on my setup. Are you setting the CDR_FLAG for all the requests? Have you also tried with different db modes? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.__com http://www.opensips- solutions.com On 04/23/2014 03:05 PM, mayamatakeshi wrote: Hello, I am trying to upgrade from opensips 1.10 (commit bfd86ab25554082053167a82655e66__9ca1c0ea7c) to 1.11 (latest 313f41ad46921d543980fa9fc71fb0__715d385085). Almost everything works. However, in my opensips.cfg I set dialog db_mode to REALTIME. And I have acc in CDR mode. Then after making some calls and confirming the dialogs were inserted in the opensips.dialog table, I kill and restart opensips. Then I terminate the calls. In 1.10, this causes proper generation of records in the acc table. However, in 1.11, no records are written to acc table. I checked the release notes for 1.11 but could not find any indication that this behavior would change: http://opensips.org/pub/__opensips/1.11.0/src/ChangeLog http://opensips.org/pub/opensips/1.11.0/src/ChangeLog Before filing a bug ticket i would like to confirm as i might be overlooking something. Here is the significant snippets of opensips.cfg: loadmodule dialog.so modparam(dialog, default_timeout, 10800) modparam(dialog, profiles_with_value, calls) modparam(dialog, db_url, mysql://opensips:opensipsrw@__localhost/opensips) modparam(dialog, db_mode, 1) # 1=REALTIME, 2=DELAYED, 3=SHUTDOWN modparam(dialog, db_update_period, 10) loadmodule acc.so modparam(acc, db_flag, DB_FLAG) modparam(acc, log_flag, LOG_FLAG) modparam(acc, log_level, -1) modparam(acc, log_facility, LOG_LOCAL0) modparam(acc, db_url, mysql://opensips:opensipsrw@__localhost/opensips) modparam(acc, failed_transaction_flag, FAILED_TRANSACTION_FLAG) modparam(acc, db_extra, domain=$avp(i:6);direction=$__avp(i:20);from_uname=$fU;to__ _uname=$tU;calling_number=$avp(__i:21);destination=$avp( i:22);__anonymous=$avp(i:23);gwid=$__avp(i:24);forward=$avp(i:25)) modparam(acc, log_extra, domain=$avp(i:6);direction=$__avp(i:20);from_uname=$fU;to__ _uname=$tU;calling_number=$avp(__i:21);destination=$avp( i:22);__anonymous=$avp(i:23);gwid=$__avp(i:24);forward=$avp(i:25)) modparam(acc, cdr_flag, CDR_FLAG) Regards, Takeshi _ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-__bin/mailman/listinfo/users http://lists.opensips.org/cgi-bin/mailman/listinfo/users _ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-__bin/mailman/listinfo/users 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
[OpenSIPS-Users] No acc records after opensips restart
Hello, I am trying to upgrade from opensips 1.10 (commit bfd86ab25554082053167a82655e669ca1c0ea7c) to 1.11 (latest 313f41ad46921d543980fa9fc71fb0715d385085). Almost everything works. However, in my opensips.cfg I set dialog db_mode to REALTIME. And I have acc in CDR mode. Then after making some calls and confirming the dialogs were inserted in the opensips.dialog table, I kill and restart opensips. Then I terminate the calls. In 1.10, this causes proper generation of records in the acc table. However, in 1.11, no records are written to acc table. I checked the release notes for 1.11 but could not find any indication that this behavior would change: http://opensips.org/pub/opensips/1.11.0/src/ChangeLog Before filing a bug ticket i would like to confirm as i might be overlooking something. Here is the significant snippets of opensips.cfg: loadmodule dialog.so modparam(dialog, default_timeout, 10800) modparam(dialog, profiles_with_value, calls) modparam(dialog, db_url, mysql://opensips:opensipsrw@localhost /opensips) modparam(dialog, db_mode, 1) # 1=REALTIME, 2=DELAYED, 3=SHUTDOWN modparam(dialog, db_update_period, 10) loadmodule acc.so modparam(acc, db_flag, DB_FLAG) modparam(acc, log_flag, LOG_FLAG) modparam(acc, log_level, -1) modparam(acc, log_facility, LOG_LOCAL0) modparam(acc, db_url, mysql://opensips:opensipsrw@localhost/opensips) modparam(acc, failed_transaction_flag, FAILED_TRANSACTION_FLAG) modparam(acc, db_extra, domain=$avp(i:6);direction=$avp(i:20);from_uname=$fU;to_uname=$tU;calling_number=$avp(i:21);destination=$avp(i:22);anonymous=$avp(i:23);gwid=$avp(i:24);forward=$avp(i:25)) modparam(acc, log_extra, domain=$avp(i:6);direction=$avp(i:20);from_uname=$fU;to_uname=$tU;calling_number=$avp(i:21);destination=$avp(i:22);anonymous=$avp(i:23);gwid=$avp(i:24);forward=$avp(i:25)) modparam(acc, cdr_flag, CDR_FLAG) Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Cancelling calls after sending '180 Ringing' via MI t_reply doesn't work
Hi Răzvan. On Tue, Dec 24, 2013 at 6:23 PM, Răzvan Crainea raz...@opensips.org wrote: Hi, Takeshi! Is the UAS sending a 100 Trying? Yes. I just confirmed now: '100 Trying' is sent by the UAS. If it's not and you are also dropping all the replies (the 180 messages), then OpenSIPS considers that no reply was received from the UAS, therefore it cannot be reached. A CANCEL would not make sense, because there is no indication that the UAS received the INVITE request. Yes, I eventually realized this would be the case and my approach was wrong. However, this doesn't explain the result of my other test: if instead of dropping it I convert 180 to 181, then the transaction is cancellable. But if after relaying this modified message I send '180 Ringing', the transaction cannot be cancelled anymore. So it looks like using MI t_reply messes with transactions. Regards, Takeshi Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 12/21/2013 03:24 AM, mayamatakeshi wrote: Hello, I have a requirement to delay the relay of '180 Ringing' for 3 seconds. So what I did was to intercept the '180 Ringing' at the onreply_route and drop it. I send the transaction_id and to-tag to an external app that starts a 3 seconds timer and uses t_reply via MI to send a '180 Ringing' on the transaction. This works fine. However, if after the delayed '180 Ringing' is sent the caller cancels the request, opensips doesn't send CANCEL to the callee. I have also tried to convert '180 Ringing' to '181 Call is Being Forwarded' instead of dropping it, and this works fine too and the call can be cancelled. However, if the delayed '180 Ringing' is sent, the call cannot be cancelled anymore. So, am I doing something wrong? Regards, Takeshi ___ 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] Cancelling calls after sending '180 Ringing' via MI t_reply doesn't work
Hello, I have a requirement to delay the relay of '180 Ringing' for 3 seconds. So what I did was to intercept the '180 Ringing' at the onreply_route and drop it. I send the transaction_id and to-tag to an external app that starts a 3 seconds timer and uses t_reply via MI to send a '180 Ringing' on the transaction. This works fine. However, if after the delayed '180 Ringing' is sent the caller cancels the request, opensips doesn't send CANCEL to the callee. I have also tried to convert '180 Ringing' to '181 Call is Being Forwarded' instead of dropping it, and this works fine too and the call can be cancelled. However, if the delayed '180 Ringing' is sent, the call cannot be cancelled anymore. So, am I doing something wrong? Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] fr_timer not working properly
On Tue, Oct 15, 2013 at 10:09 AM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, I have some servers running opensips 1.10 commit bfd86ab25554082053167a82655e669ca1c0ea7c I am seeing an issue in them that i cannot reproduce in other servers: the fr_timer is not respected. I set tm fr_timer=5 and use sipp to make a call and make opensips to send the call to an address that is not being listened by any app like 127.0.0.1:5010. In these servers, sometimes timeout happens after 6, 9, 12 seconds but never (so far after) 5 seconds. But in other machines, fr_timer is consistently respected. So I am trying to figure out what would be causing this. Does anyone have any idea of configuration of the server that could cause this? The only significant difference I see is that on the problematic servers corosync was installed for HA (to control opensips switchover) but I don't see how this could have such impact. Just in case someone else gets into this. We verified the problem only happens if the VM runs on a VM host using this processor: model name : AMD Opteron(tm) Processor 6344 On the server where we have this one: model name : Intel(R) Xeon(R) CPU X5550 @ 2.67GHz there is no problem. Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] fr_timer not working properly
On Sat, Oct 26, 2013 at 1:35 AM, Muhammad Shahzad Shafi
shah...@voip-demos.com wrote:
**
This is pretty strange, since it never happened to me on an AMD machine,
though i have different model, AMD Athlon II X2 250e.
Can you try using fr_timer_avp? e.g.
modparam(tm, fr_timer_avp, $avp(10))
modparam(tm, fr_inv_timer_avp, $avp(20))
See more info,
http://www.opensips.org/html/docs/modules/1.10.x/tm.html#id293753
Set these vars and print xlog when INVITE comes in.
branch_route {
...
$avp(10) = 5;
$avp(20) = 60;
xlog(L_NOTICE, [$pr:$fU@$si:$sp]: Setting TRY timer to '$avp(10)'
and RING timer to '$avp(20)' for call from '$fu' to '$ru' \n);
...
}
Then check in failure route,
failure_route {
...
if (t_check_status(408)) {
if ( t_local_replied(all) ) {
# local timeout with no reply received - fr_timer
xlog(L_NOTICE, [$pr:$fU@$si:$sp]: Timeout on TRY timer for
'$rm' from '$fu' to '$ru' after '$avp(10)' seconds \n);
} else if ( t_local_replied(last) ) {
# timeout with replies received - fr_inv_timer
xlog(L_NOTICE, [$pr:$fU@$si:$sp]: Timeout on RING timer
for '$rm' from '$fu' to '$ru' after '$avp(20)' seconds \n);
} else {
# received timeout
xlog(L_NOTICE, [$pr:$fU@$si:$sp]: Timeout on T2 timer for
'$rm' from '$fu' to '$ru' after '$avp(20)' seconds \n);
};
};
...
}
This will make things clearer if there is any problem.
Hello Muhammad,
thanks for the info.
I fail to make it clear that this is a problem with opensips running on
virtual machines.
The problem only happens on virtualized AMD Opteron(tm) Processor 6344 (on
KVM).
Actually, we first detected this problem using fr_timer_avp, then we
proceeded to check with fr_timer parameter.
I forgot to mention that I have opened a ticket with more details about
this:
https://github.com/OpenSIPS/opensips/issues/107
Our conclusion was that the VM host is failing to send all ticks to the VM
guest maybe assuming it is running in tickless mode (although I am using
KVM, a read about this on a VMWare doc). So time passes slowly to the VM
guest.
We could not solve yet this issue so for now, the solution is to run on a
different VM host using a different CPU model.
Regards,
Takeshi
On 2013-10-25 09:35, mayamatakeshi wrote:
On Tue, Oct 15, 2013 at 10:09 AM, mayamatakeshi
mayamatake...@gmail.comwrote:
Hello,
I have some servers running opensips 1.10 commit
bfd86ab25554082053167a82655e669ca1c0ea7c
I am seeing an issue in them that i cannot reproduce in other servers:
the fr_timer is not respected.
I set tm fr_timer=5 and use sipp to make a call and make opensips to send
the call to an address that is not being listened by any app like
127.0.0.1:5010.
In these servers, sometimes timeout happens after 6, 9, 12 seconds but
never (so far after) 5 seconds.
But in other machines, fr_timer is consistently respected.
So I am trying to figure out what would be causing this.
Does anyone have any idea of configuration of the server that could cause
this?
The only significant difference I see is that on the problematic servers
corosync was installed for HA (to control opensips switchover) but I don't
see how this could have such impact.
Just in case someone else gets into this.
We verified the problem only happens if the VM runs on a VM host using
this processor:
model name : AMD Opteron(tm) Processor 6344
On the server where we have this one:
model name : Intel(R) Xeon(R) CPU X5550 @ 2.67GHz
there is no problem.
Regards,
Takeshi
--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.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] fr_timer not working properly
Hello, I have some servers running opensips 1.10 commit bfd86ab25554082053167a82655e669ca1c0ea7c I am seeing an issue in them that i cannot reproduce in other servers: the fr_timer is not respected. I set tm fr_timer=5 and use sipp to make a call and make opensips to send the call to an address that is not being listened by any app like 127.0.0.1:5010. In these servers, sometimes timeout happens after 6, 9, 12 seconds but never (so far after) 5 seconds. But in other machines, fr_timer is consistently respected. So I am trying to figure out what would be causing this. Does anyone have any idea of configuration of the server that could cause this? The only significant difference I see is that on the problematic servers corosync was installed for HA (to control opensips switchover) but I don't see how this could have such impact. Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dialog db flush issue
Hello, I am experimenting with module dialog profiles to limit simultaneous calls from/to a subscriber. This works fine. And also works when I restart opensips: - calls established before restart are counted when limit is checked - acc record (cdr mode) is generated for the call when BYE is received. I can see this happens because when opensips shuts down gracefully, it saves dialog data to DB. Then, I experimented simulating an opensips crash by doing kill -9 pid for each opensips process. This was done after i confirmed dialog has flushed its data to DB (i am using db_mode=DELAYED and db_update_period=10). However, after restarting opensips, although I can see it was able to load dialog data: - calls established before the crash are not counted when limit is checked - no acc record is generated when BY is received. Comparing, the records for the same call that periodic dialog update generates with what a graceful shutdown saves, I can see they are different: After periodic dialog db update: mysql select * from dialog; +---+--+---+--++--+--++-+-+--+--+--+--+--++++---+++--+--+--+---+ | dlg_id| callid | from_uri | from_tag | to_uri | to_tag | mangled_from_uri | mangled_to_uri | caller_cseq | callee_cseq | caller_ping_cseq | callee_ping_cseq | caller_route_set | callee_route_set | caller_contact | callee_contact | caller_sock| callee_sock| state | start_time | timeout| vars | profiles | script_flags | flags | +---+--+---+--++--+--++-+-+--+--+--+--+--++++---+++--+--+--+---+ | 9549462304683 | b583e688-f87b-4861-936f-7dc76e33004a | sip:53535353224...@somedomain.com | d2ecbce5-9275-4788-a40c-509a23476e35 | sip:234223423423@192.168.2.177 | 39178165-9e29-4a31-af78-51e61e54920c | NULL | NULL | 0 | 21320 |0 |0 | NULL | NULL | sip:sip-tester@192.168.2.78:5060 | sip:sip-tester@192.168.2.239:10802 | udp:192.168.2.177:5060 | udp:192.168.2.177:5060 | 4 | 1380805509 | 1380816310 | NULL | NULL |0 |16 | +---+--+---+--++--+--++-+-+--+--+--+--+--++++---+++--+--+--+---+ 1 row in set (0.00 sec) After opensips graceful shutdown: mysql select * from dialog; +---+--+---+--++--+--++-+-+--+--+--+--+--++++---++++--+--+---+ | dlg_id| callid | from_uri | from_tag | to_uri |
Re: [OpenSIPS-Users] Dialog db flush issue
On Thu, Oct 3, 2013 at 10:15 PM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, I am experimenting with module dialog profiles to limit simultaneous calls from/to a subscriber. This works fine. And also works when I restart opensips: - calls established before restart are counted when limit is checked - acc record (cdr mode) is generated for the call when BYE is received. I can see this happens because when opensips shuts down gracefully, it saves dialog data to DB. Then, I experimented simulating an opensips crash by doing kill -9 pid for each opensips process. This was done after i confirmed dialog has flushed its data to DB (i am using db_mode=DELAYED and db_update_period=10). However, after restarting opensips, although I can see it was able to load dialog data: - calls established before the crash are not counted when limit is checked - no acc record is generated when BY is received. Comparing, the records for the same call that periodic dialog update generates with what a graceful shutdown saves, I can see they are different: After periodic dialog db update: mysql select * from dialog; +---+--+---+--++--+--++-+-+--+--+--+--+--++++---+++--+--+--+---+ | dlg_id| callid | from_uri | from_tag | to_uri | to_tag | mangled_from_uri | mangled_to_uri | caller_cseq | callee_cseq | caller_ping_cseq | callee_ping_cseq | caller_route_set | callee_route_set | caller_contact | callee_contact | caller_sock| callee_sock| state | start_time | timeout| vars | profiles | script_flags | flags | +---+--+---+--++--+--++-+-+--+--+--+--+--++++---+++--+--+--+---+ | 9549462304683 | b583e688-f87b-4861-936f-7dc76e33004a | sip:53535353224...@somedomain.com | d2ecbce5-9275-4788-a40c-509a23476e35 | sip:234223423423@192.168.2.177 | 39178165-9e29-4a31-af78-51e61e54920c | NULL | NULL | 0 | 21320 |0 |0 | NULL | NULL | sip:sip-tester@192.168.2.78:5060 | sip:sip-tester@192.168.2.239:10802 | udp:192.168.2.177:5060 | udp:192.168.2.177:5060 | 4 | 1380805509 | 1380816310 | NULL | NULL |0 |16 | +---+--+---+--++--+--++-+-+--+--+--+--+--++++---+++--+--+--+---+ 1 row in set (0.00 sec) After opensips graceful shutdown: mysql select * from dialog; +---+--+---+--++--+--++-+-+--+--+--+--+--++++---++++--+--+---+ | dlg_id| callid
Re: [OpenSIPS-Users] Stats for module dialog using binary interface
Hi Liviu, understood. I failed to notice 13 + 20 = 33. Thanks for the clarification. Regards, Takeshi On Tue, Sep 10, 2013 at 7:01 PM, Liviu Chircu li...@opensips.org wrote: Hello Takeshi, As you probably noticed, the difference comes from several failed dialogs (most likely from a negative reply). Since only confirmed dialogs are replicated, the standby server will not receive all statistics. Although the issue is harmless, to me it looks like it needs a fix. I'll put in on my TODO list. Best regards, Liviu Chircu OpenSIPS Devhttp://www.opensips-solutions.com On 09/09/2013 03:29 PM, mayamatakeshi wrote: I am experimenting with the binary interface to replicate dialog data between 2 instances of opensips After a batch of tests (done after restarting opensips), I compared the stats for module dialog and verified they are not the same: Active server: # opensipsctl fifo get_statistics dialog: dialog:active_dialogs = 0 dialog:early_dialogs = 0 dialog:processed_dialogs = 33 dialog:expired_dialogs = 0 dialog:failed_dialogs = 20 dialog:create_sent = 13 dialog:update_sent = 49 dialog:delete_sent = 48 dialog:create_recv = 0 dialog:update_recv = 0 dialog:delete_recv = 0 Standby server: # opensipsctl fifo get_statistics dialog: dialog:active_dialogs = 0 dialog:early_dialogs = 0 dialog:processed_dialogs = 13 dialog:expired_dialogs = 0 dialog:failed_dialogs = 0 dialog:create_sent = 0 dialog:update_sent = 0 dialog:delete_sent = 0 dialog:create_recv = 13 dialog:update_recv = 49 dialog:delete_recv = 48 Is the above expected? (obs: repeating the tests always produce the same numbers). Regards, Takeshi ___ 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] Stats for module dialog using binary interface
I am experimenting with the binary interface to replicate dialog data between 2 instances of opensips After a batch of tests (done after restarting opensips), I compared the stats for module dialog and verified they are not the same: Active server: # opensipsctl fifo get_statistics dialog: dialog:active_dialogs = 0 dialog:early_dialogs = 0 dialog:processed_dialogs = 33 dialog:expired_dialogs = 0 dialog:failed_dialogs = 20 dialog:create_sent = 13 dialog:update_sent = 49 dialog:delete_sent = 48 dialog:create_recv = 0 dialog:update_recv = 0 dialog:delete_recv = 0 Standby server: # opensipsctl fifo get_statistics dialog: dialog:active_dialogs = 0 dialog:early_dialogs = 0 dialog:processed_dialogs = 13 dialog:expired_dialogs = 0 dialog:failed_dialogs = 0 dialog:create_sent = 0 dialog:update_sent = 0 dialog:delete_sent = 0 dialog:create_recv = 13 dialog:update_recv = 49 dialog:delete_recv = 48 Is the above expected? (obs: repeating the tests always produce the same numbers). Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Accepting calls only form Some IP Address
Try this:
if ($si == 192.168.0.35) {
t_relay();
} else {
sl_send_reply(403, Forbidden);
exit;
}
regards,
Takeshi
On Fri, Aug 30, 2013 at 7:39 PM, Danny Dias ing.diasda...@gmail.com wrote:
sorry, my small script is like this:
if (check_address(0,192.168.0.35,0,ANY,, )) {
t_relay();
}
else {
sl_send_reply(403, Forbidden);
exit;
}
Still receiving the 403
2013/8/30 Danny Dias ing.diasda...@gmail.com
Finally,
I did it like this:
if (!check_address(0,192.168.0.35,0,ANY,, )) {
t_relay();
}
else {
sl_send_reply(403, Forbidden);
exit;
}
But now, all the calls from source ip address 192.168.0.35 receive a 403:
U 192.168.0.35:3832 - 192.168.1.20:5060
INVITE sip:1001@192.168.1.20 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.35:3832
;branch=z9hG4bK-d8754z-c4f09d265010a511-1---d8754z-;rport.
Max-Forwards: 70.
Contact: sip:1000@192.168.0.35:3832.
To: sip:1001@192.168.1.20.
From: 1000sip:1000@192.168.1.20;tag=6be1ed13.
Call-ID: ZGZiMjEwMzFjZjRkNDFmMzdhZGJkNTgxYzlmYjE4MTY.
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: Bria 3 release 3.5.3 stamp 70600.
Content-Length: 208.
.
v=0.
o=- 1377858559536092 1 IN IP4 192.168.0.35.
s=Bria 3 release 3.5.3 stamp 70600.
c=IN IP4 192.168.0.35.
t=0 0.
m=audio 53766 RTP/AVP 8 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
U 192.168.1.20:5060 - 192.168.0.35:3832
SIP/2.0 403 Forbidden.
Via: SIP/2.0/UDP 192.168.0.35:3832
;received=192.168.0.35;branch=z9hG4bK-d8754z-c4f09d265010a511-1---d8754z-;rport=3832.
To: sip:1001@192.168.1.20;tag=7ff88f74df89822193682e3f23116cc8.59de.
From: 1000sip:1000@192.168.1.20;tag=6be1ed13.
Call-ID: ZGZiMjEwMzFjZjRkNDFmMzdhZGJkNTgxYzlmYjE4MTY.
CSeq: 1 INVITE.
Server: OpenSIPS (1.9.1-notls (i386/linux)).
Content-Length: 0.
2013/8/30 Danny Dias ing.diasda...@gmail.com
So,
It would be something like this (not quite sure...):
route {
if (check_address(0,192.168.2.135,0,ANY,, )) {
t_relay();
else if (sl_send_reply(403, Forbidden));
exit;
}
.
.
In this small script at the very begining of route script, i will only
permit calls from ip 192.168.2.135, any other ip, will receive a 403. Am
i right?
Thanks
2013/8/30 Víctor Fernández Martínez vfernan...@barracuda.com
Hi Danny,
You can check the source IP address using $si and send a forbidden
response if
it doesn't match your needs.
Best regards.
On Friday 30 August 2013 03:48:25 Danny Días wrote:
Hi,
I need to configure OpenSIPS to only accept calls from 1 IP address
and
without any kind of authentication (no password required for
clients); i
wonder if there's a function in OpenSIPS to accomplish this?
Thanks so much
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
*SIP:* da...@voice.danntel.net http://www.danntel.net/?page_id=189
*Web: *http://www.danntel.net
--
*SIP:* da...@voice.danntel.net http://www.danntel.net/?page_id=189
*Web: *http://www.danntel.net
--
*SIP:* da...@voice.danntel.net http://www.danntel.net/?page_id=189
*Web: *http://www.danntel.net
___
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-Devel] [NEW] Dialog replication using a new core interface
Sorry, please ignore. Replied to the wrong post. On Wed, Aug 7, 2013 at 9:54 AM, mayamatakeshi mayamatake...@gmail.comwrote: Hello Vlad, I tested and confirmed it is OK now. Thanks. Regards, Takeshi On Wed, Aug 7, 2013 at 4:52 AM, Vlad Paiu vladp...@opensips.org wrote: Hello, What are you thinking about more exactly here in terms of security ? The OpenSIPS worker processes that listen for BIN replicated packages do not perform any IP authentication by themselves, so if you leave those UDP ports open from the outside, you are leaving yourself exposed to outside attackers coming in and either deleting some of the existing dialogs ( by sending you some binary packages that destroy an ongoing dialog ) or filling up your shared memory ( by sending you 'new dialog' binary packages ). Currently, it's left to the OpenSIPS administrator to properly configure the firewall so that the binary interface listeners ( the ones specified by bin_listen=127.0.0.1: ) are only open for the other OpenSIPS instance IPs. Best Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.**com http://www.opensips-solutions.com On 08/01/2013 07:58 PM, Nick Khamis wrote: What needs to be considered in terms of security? Nick. On 7/31/13, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Ryan, This has nothing to do with dialog pining or accounting - the new interface allows OpenSIPS to replicate the dialog state to another OpenSIPS instance. If I misunderstood you, please rephrase :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.**com http://www.opensips-solutions.com On 07/29/2013 08:20 PM, Ryan Bullock wrote: This is pretty exciting! What are the plans for how this will work with features such as dialog pinging and accounting? Regards, Ryan On Mon, Jul 29, 2013 at 9:46 AM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: In long term we plan to use the BIN interface to replicate even more internal data between multiple OpenSIPS instances, like doing registration replication (instead of doing it from script via SIP). Theoretically it may be used for replicating even transaction state between 2 OpenSIPS instances - imagine having a call ringing on instance A and being accepted on instance B (after a failover) - 0% losses ! Aside realtime data replication, the BIN interface is to be used also for exchanging any other type of information between OpenSIPS instances, like federating multiple instances. The main advantages of the BIN interface over the MI interface : - BIN is binary encoded so much faster (as performance) - BIN interface has both sender and receiver in OpenSIPS (MI has only the receiver) - MI is for external usage, while BIN is internal (opensips2opensips) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.**comhttp://www.opensips-solutions.com On 07/29/2013 06:22 PM, Liviu Chircu wrote: Hello all, OpenSIPS just got better with a /new core interface/ and a /new failover mechanism/! The purpose of the new *Binary Internal Interface *is to offer a fast and efficient communication channel between OpenSIPS instances. OpenSIPS modules can now use this core interface to send/receive packets with specific information. A common usage case for this feature would be data replication between a primary instance and a backup one. This is especially useful in scenarios with OpenSIPS instances which handle large amounts of concurrent calls, so that failover through a database backend is not feasible anymore due to the significant time required in order to load the needed tables. As an example of using the interface, the dialog module now offers the possibility of *replicating dialogs* to another instance. The script writer may now configure a set of proxies which will receive dialog-related events: /creation/, /confirmation/ and /deletion/, all in /realtime/. These messages are compact and they are sent over UDP. The dialog module now also exports several new statistics which show the total sent/received replication packets. Configuring UDP listeners for the new interface is trivial and explained in the OpenSIPS manuals [1]. [1]: http://www.opensips.org/**Documentation/Interface-Binaryhttp://www.opensips.org/Documentation/Interface-Binary Best regards, -- Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.**comhttp://www.opensips-solutions.com __**_ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.**orgUsers
Re: [OpenSIPS-Users] How to get is_from_gw to fill gw_id_avp
Hello Vlad, I tested and confirmed it is OK now. Thanks. Regards, Takeshi On Wed, Aug 7, 2013 at 4:33 AM, Vlad Paiu vladp...@opensips.org wrote: Hello, Actually, the ERROR:core:fixup_uint: bad number -1 error message was a bug in the code, that it wouldn't accept negative numbers for the gateway type ( as the docs suggest would match any gw type ). I've just pushed a fix for this on 1.8, 1.9 and 1.10. Best Regards, Vlad Paiu OpenSIPS Developerhttp://www.opensips-solutions.com On 08/04/2013 11:39 PM, mayamatakeshi wrote: Thanks, actually, I realized it was my mistake: i was calling is_from_gw more than once but only used i on one place and so opensips was not always filling the AVP as it was not requested. regards, Takeshi On Mon, Aug 5, 2013 at 9:24 AM, Laszlo las...@voipfreak.net wrote: http://openser.org/pipermail/users/2013-May/025705.html maybe this can give you an idea.. -Laszlo 2013/8/3 mayamatakeshi mayamatake...@gmail.com Hello, can anyone give me an example of how to call drouting function is_from_gw ? http://www.opensips.org/html/docs/modules/1.9.x/drouting.html#id293958 The doc gives this example: if (is_from_gw(3,1) which seems to be incorrect as 1 is not a valid flag. I am trying to use flag i to get the gwid in the AVP set using gw_id_avp but it is not working. I already tried as: if (is_from_gw(, i)) and if (is_from_gw(0, i)) but it is not working: the avp set by avp_gw_id is no filled. I know my configuration is correct because do_routing() does fill the avp. Obs: the doc also says that if I don't care about the type of the gateway I can pass a negative number. So I tried: if (is_from_gw(-1, i) But then i get: Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: ERROR:core:fixup_uint: bad number -1 Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: ERROR:core:fix_actions: fixing failed (code=-6) at cfg line 227 Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: CRITICAL:core:fix_expr: fix_actions error Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: ERROR:core:main: failed to fix configuration with err code -6 In case this is a bug, I tried with this git commit (head): commit 70b6629f88e2e3fc51e42154b7db8fb7b94417a1 Author: Damien Sandras dsand...@beip.be Date: Wed Jul 10 17:59:34 2013 +0200 regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- Kind regards, Laszlo Bekesi http://voipfreak.net ___ 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-Devel] [NEW] Dialog replication using a new core interface
Hello Vlad, I tested and confirmed it is OK now. Thanks. Regards, Takeshi On Wed, Aug 7, 2013 at 4:52 AM, Vlad Paiu vladp...@opensips.org wrote: Hello, What are you thinking about more exactly here in terms of security ? The OpenSIPS worker processes that listen for BIN replicated packages do not perform any IP authentication by themselves, so if you leave those UDP ports open from the outside, you are leaving yourself exposed to outside attackers coming in and either deleting some of the existing dialogs ( by sending you some binary packages that destroy an ongoing dialog ) or filling up your shared memory ( by sending you 'new dialog' binary packages ). Currently, it's left to the OpenSIPS administrator to properly configure the firewall so that the binary interface listeners ( the ones specified by bin_listen=127.0.0.1: ) are only open for the other OpenSIPS instance IPs. Best Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.**com http://www.opensips-solutions.com On 08/01/2013 07:58 PM, Nick Khamis wrote: What needs to be considered in terms of security? Nick. On 7/31/13, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Ryan, This has nothing to do with dialog pining or accounting - the new interface allows OpenSIPS to replicate the dialog state to another OpenSIPS instance. If I misunderstood you, please rephrase :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.**com http://www.opensips-solutions.com On 07/29/2013 08:20 PM, Ryan Bullock wrote: This is pretty exciting! What are the plans for how this will work with features such as dialog pinging and accounting? Regards, Ryan On Mon, Jul 29, 2013 at 9:46 AM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: In long term we plan to use the BIN interface to replicate even more internal data between multiple OpenSIPS instances, like doing registration replication (instead of doing it from script via SIP). Theoretically it may be used for replicating even transaction state between 2 OpenSIPS instances - imagine having a call ringing on instance A and being accepted on instance B (after a failover) - 0% losses ! Aside realtime data replication, the BIN interface is to be used also for exchanging any other type of information between OpenSIPS instances, like federating multiple instances. The main advantages of the BIN interface over the MI interface : - BIN is binary encoded so much faster (as performance) - BIN interface has both sender and receiver in OpenSIPS (MI has only the receiver) - MI is for external usage, while BIN is internal (opensips2opensips) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.**comhttp://www.opensips-solutions.com On 07/29/2013 06:22 PM, Liviu Chircu wrote: Hello all, OpenSIPS just got better with a /new core interface/ and a /new failover mechanism/! The purpose of the new *Binary Internal Interface *is to offer a fast and efficient communication channel between OpenSIPS instances. OpenSIPS modules can now use this core interface to send/receive packets with specific information. A common usage case for this feature would be data replication between a primary instance and a backup one. This is especially useful in scenarios with OpenSIPS instances which handle large amounts of concurrent calls, so that failover through a database backend is not feasible anymore due to the significant time required in order to load the needed tables. As an example of using the interface, the dialog module now offers the possibility of *replicating dialogs* to another instance. The script writer may now configure a set of proxies which will receive dialog-related events: /creation/, /confirmation/ and /deletion/, all in /realtime/. These messages are compact and they are sent over UDP. The dialog module now also exports several new statistics which show the total sent/received replication packets. Configuring UDP listeners for the new interface is trivial and explained in the OpenSIPS manuals [1]. [1]: http://www.opensips.org/**Documentation/Interface-Binaryhttp://www.opensips.org/Documentation/Interface-Binary Best regards, -- Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.**comhttp://www.opensips-solutions.com __**_ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.**orgUsers@lists.opensips.org http://lists.opensips.org/cgi-**bin/mailman/listinfo/usershttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to get is_from_gw to fill gw_id_avp
Thanks, actually, I realized it was my mistake: i was calling is_from_gw more than once but only used i on one place and so opensips was not always filling the AVP as it was not requested. regards, Takeshi On Mon, Aug 5, 2013 at 9:24 AM, Laszlo las...@voipfreak.net wrote: http://openser.org/pipermail/users/2013-May/025705.html maybe this can give you an idea.. -Laszlo 2013/8/3 mayamatakeshi mayamatake...@gmail.com Hello, can anyone give me an example of how to call drouting function is_from_gw ? http://www.opensips.org/html/docs/modules/1.9.x/drouting.html#id293958 http://www.opensips.org/html/docs/modules/devel/drouting.html#id294786 The doc gives this example: if (is_from_gw(3,1) which seems to be incorrect as 1 is not a valid flag. I am trying to use flag i to get the gwid in the AVP set using gw_id_avp but it is not working. I already tried as: if (is_from_gw(, i)) and if (is_from_gw(0, i)) but it is not working: the avp set by avp_gw_id is no filled. I know my configuration is correct because do_routing() does fill the avp. Obs: the doc also says that if I don't care about the type of the gateway I can pass a negative number. So I tried: if (is_from_gw(-1, i) But then i get: Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: ERROR:core:fixup_uint: bad number -1 Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: ERROR:core:fix_actions: fixing failed (code=-6) at cfg line 227 Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: CRITICAL:core:fix_expr: fix_actions error Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: ERROR:core:main: failed to fix configuration with err code -6 In case this is a bug, I tried with this git commit (head): commit 70b6629f88e2e3fc51e42154b7db8fb7b94417a1 Author: Damien Sandras dsand...@beip.be Date: Wed Jul 10 17:59:34 2013 +0200 regards, Takeshi http://www.opensips.org/html/docs/modules/devel/drouting.html#id294786 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- Kind regards, Laszlo Bekesi http://voipfreak.net ___ 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] How to get is_from_gw to fill gw_id_avp
Hello, can anyone give me an example of how to call drouting function is_from_gw ? http://www.opensips.org/html/docs/modules/1.9.x/drouting.html#id293958 http://www.opensips.org/html/docs/modules/devel/drouting.html#id294786 The doc gives this example: if (is_from_gw(3,1) which seems to be incorrect as 1 is not a valid flag. I am trying to use flag i to get the gwid in the AVP set using gw_id_avp but it is not working. I already tried as: if (is_from_gw(, i)) and if (is_from_gw(0, i)) but it is not working: the avp set by avp_gw_id is no filled. I know my configuration is correct because do_routing() does fill the avp. Obs: the doc also says that if I don't care about the type of the gateway I can pass a negative number. So I tried: if (is_from_gw(-1, i) But then i get: Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: ERROR:core:fixup_uint: bad number -1 Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: ERROR:core:fix_actions: fixing failed (code=-6) at cfg line 227 Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: CRITICAL:core:fix_expr: fix_actions error Aug 3 16:53:44 SERVER /usr/local/src/git/opensips-head/opensips[1454]: ERROR:core:main: failed to fix configuration with err code -6 In case this is a bug, I tried with this git commit (head): commit 70b6629f88e2e3fc51e42154b7db8fb7b94417a1 Author: Damien Sandras dsand...@beip.be Date: Wed Jul 10 17:59:34 2013 +0200 regards, Takeshi http://www.opensips.org/html/docs/modules/devel/drouting.html#id294786 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Reception of 100 Trying doesn't cancel tm fr_timer
On Mon, Jul 15, 2013 at 11:49 AM, mayamatakeshi mayamatake...@gmail.comwrote: On Mon, Jul 15, 2013 at 10:13 AM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, we are using opensips 1.6 r. 7055 under CentOS 5.3 x86_64 and this works fine on several machines. I am tasked with preparing upgrade to CentOS 6.4 x86_64 so I have prepared a Virtual Machine to run this OS, installed opensips and used the same opensips.cfg . However, I am seeing the following issue: reception of 100 Trying doesn't cancel module tm fr_timer but if other provisional responses like 180 Ringing is received, fr_timer is canceled. I am sure the 100 Trying is received by opensips because it doesn't retransmit the INVITE. Sorry, by mistake I looked at the wrong capture. Indeed, opensips is retransmitting the INVITE. But checking Call-ID, from-tag and Via headers, they match and so, opensips should not do retransmission. It seems somehow it is ignoring the 100 Trying. So to further investigate I enabled debug with opensipsctl fifo debug 4 (instead of default debug 0), but surprisingly, now opensips accepts the 180 Trying (doesn't retransmit the INVITE) and fr_timer is canceled. In the above I meant 100 Trying. I tested several times and just by switching debug off and on is enough to cause the problem to appear/disappear. Everything else seems to be fine. Does anyone has any idea why this could be happening? Also, to confirm I have prepared 3 CentOS 6.4 x86_64 VMs, but in one of them the problem doesn't happen but I don't know why this is so as i compared the packages installed (yum list installed) and they are all the same, including version numbers. Also, on the machine where the problem doesn't manifest itself, it is the opposite: if I enable debug with opensipsctl fifo debug 4, fr_timer fails to be canceled by 100 Trying. Retrying several times, the problem occasionally happens on this machine too. So it seems to be some race condition. I was able to try with latest revisions of 1.7 and 1.8 branches and the problem also happens with them. (Also I have tried upgrading to latest branch 1.6 r. 10024, and the problem persists). Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Reception of 100 Trying doesn't cancel tm fr_timer
On Mon, Jul 15, 2013 at 7:56 PM, mayamatakeshi mayamatake...@gmail.comwrote: On Mon, Jul 15, 2013 at 11:49 AM, mayamatakeshi mayamatake...@gmail.comwrote: On Mon, Jul 15, 2013 at 10:13 AM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, we are using opensips 1.6 r. 7055 under CentOS 5.3 x86_64 and this works fine on several machines. I am tasked with preparing upgrade to CentOS 6.4 x86_64 so I have prepared a Virtual Machine to run this OS, installed opensips and used the same opensips.cfg . However, I am seeing the following issue: reception of 100 Trying doesn't cancel module tm fr_timer but if other provisional responses like 180 Ringing is received, fr_timer is canceled. I am sure the 100 Trying is received by opensips because it doesn't retransmit the INVITE. Sorry, by mistake I looked at the wrong capture. Indeed, opensips is retransmitting the INVITE. But checking Call-ID, from-tag and Via headers, they match and so, opensips should not do retransmission. It seems somehow it is ignoring the 100 Trying. So to further investigate I enabled debug with opensipsctl fifo debug 4 (instead of default debug 0), but surprisingly, now opensips accepts the 180 Trying (doesn't retransmit the INVITE) and fr_timer is canceled. In the above I meant 100 Trying. I tested several times and just by switching debug off and on is enough to cause the problem to appear/disappear. Everything else seems to be fine. Does anyone has any idea why this could be happening? Also, to confirm I have prepared 3 CentOS 6.4 x86_64 VMs, but in one of them the problem doesn't happen but I don't know why this is so as i compared the packages installed (yum list installed) and they are all the same, including version numbers. Also, on the machine where the problem doesn't manifest itself, it is the opposite: if I enable debug with opensipsctl fifo debug 4, fr_timer fails to be canceled by 100 Trying. Retrying several times, the problem occasionally happens on this machine too. So it seems to be some race condition. I was able to try with latest revisions of 1.7 and 1.8 branches and the problem also happens with them. In my tests, I have this scenario: sipp uac -- INVITE -- opensips -- FreeSWITCH sipp uac -- opensips -- 100 Trying -- FreeSWITCH however, if I replace FreeSWITCH with a sipp uas, the problem doesn't happen. I took care of preparing the sipp uas to generate the 100 Trying exactly the same as FS would do it, but still I could not reproduce the problem. So i think the presence of FS has some strange influence in the environment and i cannot be sure the problem is with opensips itself anymore. Sorry for the noise. Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Reception of 100 Trying doesn't cancel tm fr_timer
Hello, we are using opensips 1.6 r. 7055 under CentOS 5.3 x86_64 and this works fine on several machines. I am tasked with preparing upgrade to CentOS 6.4 x86_64 so I have prepared a Virtual Machine to run this OS, installed opensips and used the same opensips.cfg . However, I am seeing the following issue: reception of 100 Trying doesn't cancel module tm fr_timer but if other provisional responses like 180 Ringing is received, fr_timer is canceled. I am sure the 100 Trying is received by opensips because it doesn't retransmit the INVITE. Everything else seems to be fine. Does anyone has any idea why this could be happening? Also, to confirm I have prepared 3 CentOS 6.4 x86_64 VMs, but in one of them the problem doesn't happen but I don't know why this is so as i compared the packages installed (yum list installed) and they are all the same, including version numbers. (Also I have tried upgrading to latest branch 1.6 r. 10024, and the problem persists). Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Reception of 100 Trying doesn't cancel tm fr_timer
On Mon, Jul 15, 2013 at 10:13 AM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, we are using opensips 1.6 r. 7055 under CentOS 5.3 x86_64 and this works fine on several machines. I am tasked with preparing upgrade to CentOS 6.4 x86_64 so I have prepared a Virtual Machine to run this OS, installed opensips and used the same opensips.cfg . However, I am seeing the following issue: reception of 100 Trying doesn't cancel module tm fr_timer but if other provisional responses like 180 Ringing is received, fr_timer is canceled. I am sure the 100 Trying is received by opensips because it doesn't retransmit the INVITE. Sorry, by mistake I looked at the wrong capture. Indeed, opensips is retransmitting the INVITE. But checking Call-ID, from-tag and Via headers, they match and so, opensips should not do retransmission. It seems somehow it is ignoring the 100 Trying. So to further investigate I enabled debug with opensipsctl fifo debug 4 (instead of default debug 0), but surprisingly, now opensips accepts the 180 Trying (doesn't retransmit the INVITE) and fr_timer is canceled. I tested several times and just by switching debug off and on is enough to cause the problem to appear/disappear. Everything else seems to be fine. Does anyone has any idea why this could be happening? Also, to confirm I have prepared 3 CentOS 6.4 x86_64 VMs, but in one of them the problem doesn't happen but I don't know why this is so as i compared the packages installed (yum list installed) and they are all the same, including version numbers. Also, on the machine where the problem doesn't manifest itself, it is the opposite: if I enable debug with opensipsctl fifo debug 4, fr_timer fails to be canceled by 100 Trying. (Also I have tried upgrading to latest branch 1.6 r. 10024, and the problem persists). Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] For what do you use tm MI function t_reply?
Hello, out of curiosity, for what scenario the tm MI function t_reply was envisioned for? And how would I get the trans_id necessary to use it? Regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] For what do you use tm MI function t_reply?
Hello, I have found the explanation while reading old posts. On Mon, Apr 30, 2012 at 12:13 AM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, out of curiosity, for what scenario the tm MI function t_reply was envisioned for? It can be used to have some control of calls by an external entity (not sure of what people are doing with this right now, but I am sure there are good applications for it). And how would I get the trans_id necessary to use it? On the cfg file, by using functions t_write_req(info,fifo) and t_write_unix(info,sock) Here's a dump of what these would write: 0.3 sip_request.calling_useragent INVITE user2 192.168.0.129 127.0.0.1 . sip:user2@192.168.0.129:5061 sip:mod_sofia@127.0.0.1:5070 displayname.user1 sip:us...@test1.com sip:user2@127.0.0.1 574adad2-0cc2-1230-e9b9-000c29db16eb vjaSNUXZB52DN 27523925 46961:1279687637 . . P-MsgFlags: 1 the_fu: sip:us...@test1.com . v=0 o=FreeSWITCH 1335707806 1335707807 IN IP4 192.168.2.9 s=FreeSWITCH c=IN IP4 192.168.2.9 t=0 0 m=audio 12172 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 From the looks, the trans_id is the 16th line 46961:1279687637. Regards, Takeshi. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] creating a regular expression
On Mon, Oct 10, 2011 at 9:20 PM, Toyima Dias toyim...@gmail.com wrote: sorry, the range is 31297 - 31336... 2011/10/10 Toyima Dias toyim...@gmail.com Hello, i would like some help on a regular expression using dialplan module and regular expressions...this is what i want: construct a regular expression for the following range: 31297-313336...i heve this simple string ^31[2-3]the problem is the range from 97 to 36...any help please? I don't know if there is a simpler way. Here' it what i would use: ^(3129[789]|313([0-2][0-9]|3[0-6]))$ Test with this: #!/usr/bin/python import re r = re.compile(^(3129[789]|313([0-2][0-9]|3[0-6]))$) count = 0 for i in range(10): if r.match(str(i)): count = count + 1 print Expected: + str(31336 - 31296) print Result:+ str(count) regards, Takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] creating a regular expression
On Mon, Oct 10, 2011 at 9:40 PM, mayamatakeshi mayamatake...@gmail.comwrote: On Mon, Oct 10, 2011 at 9:20 PM, Toyima Dias toyim...@gmail.com wrote: sorry, the range is 31297 - 31336... 2011/10/10 Toyima Dias toyim...@gmail.com Hello, i would like some help on a regular expression using dialplan module and regular expressions...this is what i want: construct a regular expression for the following range: 31297-313336...i heve this simple string ^31[2-3]the problem is the range from 97 to 36...any help please? I don't know if there is a simpler way. Here' it what i would use: ^(3129[789]|313([0-2][0-9]|3[0-6]))$ Test with this: #!/usr/bin/python import re r = re.compile(^(3129[789]|313([0-2][0-9]|3[0-6]))$) count = 0 for i in range(10): if r.match(str(i)): count = count + 1 print Expected: + str(31336 - 31296) print Result:+ str(count) And be aware that there are some variations in regex syntax between tools. So this might not work the same with opensips. You need to test it. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parameter port not found in module mi_xmlrpc
On Tue, Apr 12, 2011 at 4:57 PM, Toyima Dias toyim...@gmail.com wrote: Hello, I'm receiving the following errors on my opensips.log: *ERROR:core:set_mod_param_regex: parameter port not found in module mi_xmlrpc * This is what i've added to my opensips.cfg: loadmodule mi_xmlrpc.so loadmodule siptrace.so # - setting module-specific parameters --- MODPARAMS FRO SIP TRACE modparam(mi_xmlrpc,port, 8000) Have you tried: modparam(mi_xmlrpc,port, 8000) modparam(siptrace, db_url, mysql://opensips:opensipsrw@localhost/opensips) modparam(siptrace, trace_flag, 22) modparam(siptrace, traced_user_avp, $avp(s:traceuser)) What could be wrong? i'm using *Server:: OpenSIPS (1.6.4-2-notls (i386/linux))* ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] BYE processing and AVP
On Tue, Nov 30, 2010 at 10:07 PM, Anton Zagorskiy
a.zagors...@oyster-telecom.ru wrote:
Hello.
During INVITE processing I'm setting up some avp variables. When I'm
receiving a BYE request all avps are null. Why does this happen?
A part of the script:
if (has_totag())
{
if (loose_route())
{
if (is_method(BYE))
{
xlog(*** BYE $avp(i:2));
From the docs (http://www.opensips.org/Resources/DocsCoreVar#toc2):
AVPs are part of the transaction context, so they will be visible
everywhere where the transaction is present.
br,
takeshi
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] BYE processing and AVP
2010/11/30 Anton Zagorskiy a.zagors...@oyster-telecom.ru
Why BYE isn't a part of the transaction?
From RFC3261:
Specifically, a SIP transaction consists of a single request and any
responses to
that request, which include zero or more provisional responses and
one or more final responses.
But when opensips docs say transaction, it is also implying that you are
using module tm (if you don't use tm functions, then you are doing stateless
processing and no AVP will be saved).
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of mayamatakeshi
Sent: Tuesday, November 30, 2010 4:13 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] BYE processing and AVP
On Tue, Nov 30, 2010 at 10:07 PM, Anton Zagorskiy
a.zagors...@oyster-telecom.ru wrote:
Hello.
During INVITE processing I'm setting up some avp variables. When I'm
receiving a BYE request all avps are null. Why does this happen?
A part of the script:
if (has_totag())
{
if (loose_route())
{
if (is_method(BYE))
{
xlog(*** BYE $avp(i:2));
From the docs (http://www.opensips.org/Resources/DocsCoreVar#toc2):
AVPs are part of the transaction context, so they will be visible
everywhere where the transaction is present.
br,
takeshi
___
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] BYE processing and AVP
On Tue, Nov 30, 2010 at 11:19 PM, mayamatakeshi mayamatake...@gmail.comwrote:
2010/11/30 Anton Zagorskiy a.zagors...@oyster-telecom.ru
Why BYE isn't a part of the transaction?
From RFC3261:
Specifically, a SIP transaction consists of a single request and any
responses to
that request, which include zero or more provisional responses and
one or more final responses.
But when opensips docs say transaction, it is also implying that you are
using module tm (if you don't use tm functions, then you are doing stateless
processing and no AVP will be saved).
For completeness,
the BYE is part of the dialog established by the INVITE transaction that
ended when 200 OK was received. But the BYE itself starts another
transaction.
So, the AVPs you set in the processing of the INVITE will be available (as
long as you use module tm) when your process responses for that INVITE (180,
183, 200 etc), but not when you are handling other requests in the dialog.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of mayamatakeshi
Sent: Tuesday, November 30, 2010 4:13 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] BYE processing and AVP
On Tue, Nov 30, 2010 at 10:07 PM, Anton Zagorskiy
a.zagors...@oyster-telecom.ru wrote:
Hello.
During INVITE processing I'm setting up some avp variables. When I'm
receiving a BYE request all avps are null. Why does this happen?
A part of the script:
if (has_totag())
{
if (loose_route())
{
if (is_method(BYE))
{
xlog(*** BYE $avp(i:2));
From the docs (http://www.opensips.org/Resources/DocsCoreVar#toc2):
AVPs are part of the transaction context, so they will be visible
everywhere where the transaction is present.
br,
takeshi
___
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] Forcing minimum value for SUBSCRIBE Expires
I can see there is a parameter max_expires_subscribe for module presence, but there is no min_expires_subscribe. Is it possible to force a minimum value for SUBSCRIBE expires? br, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How opensis can manage different ports.
On Sat, Sep 18, 2010 at 7:57 AM, Stefano Pisani
stefano.pis...@omnianet.itwrote:
No I don't.
I want that opensis uses same port of registration for INVITE.
i.e. the client is register on port 5061, opensips uses from port 5061
for INVITE
If you are dealing with more than 2 ports, this might get complicated.
But in case of just 2 ports, you could check the ReceivedPort and set a
bflag during handling of the REGISTER request:
if($Rp == 5060) {
setbflag(BFLAG_RECEIVED_ON_PORT_5060);
}
The bflags are saved with the registration, so you can check them in a
branch_route called after lookup and force the port to be used to relay the
request:
if (lookup(location)) {
t_on_branch(USER_BRANCH);
if (!t_relay()) {
sl_reply_error();
exit;
}
}
branch_route [USER_BRANCH] {
if (isbflagset(BFLAG_RECEIVED_ON_PORT_5060)) {
force_send_socket(MY_IP_ADDRESS:5060);
} else {
force_send_socket(MY_IP_ADDRESS:5061);
}
}
(obs: I actually, never tested this. I'm writing from my recollections of
similar problems I had to deal with).
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How opensis can manage different ports.
On Sat, Sep 18, 2010 at 11:49 AM, mayamatakeshi mayamatake...@gmail.comwrote:
On Sat, Sep 18, 2010 at 7:57 AM, Stefano Pisani
stefano.pis...@omnianet.it wrote:
No I don't.
I want that opensis uses same port of registration for INVITE.
i.e. the client is register on port 5061, opensips uses from port 5061
for INVITE
If you are dealing with more than 2 ports, this might get complicated.
But in case of just 2 ports, you could check the ReceivedPort and set a
bflag during handling of the REGISTER request:
if($Rp == 5060) {
setbflag(BFLAG_RECEIVED_ON_PORT_5060);
}
The bflags are saved with the registration, so you can check them in a
branch_route called after lookup and force the port to be used to relay the
request:
if (lookup(location)) {
t_on_branch(USER_BRANCH);
if (!t_relay()) {
sl_reply_error();
exit;
}
}
branch_route [USER_BRANCH] {
if (isbflagset(BFLAG_RECEIVED_ON_PORT_5060)) {
force_send_socket(MY_IP_ADDRESS:5060);
} else {
force_send_socket(MY_IP_ADDRESS:5061);
}
}
(obs: I actually, never tested this. I'm writing from my recollections of
similar problems I had to deal with).
I looked at the docs. I'm not sure but maybe, the socket where the message
was received will be available in the branch_route as $(branch(socket)):
http://www.opensips.org/Resources/DocsCoreVar16#toc16
At least I can see this info being stored in the location table and I expect
the branch processing to load it as it does with the bflags.
So maybe you can use just $(branch(socket)) in the call to
force_send_socket:
force_send_socket($(branch(socket)));
(I don't know if it is necessary to remove the sip: prefix from the socket
info).
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Multiple contact entries
On Thu, Aug 5, 2010 at 2:25 AM, David Cunningham dcunning...@voisonics.comwrote: Hello, We have a system using the usrloc module (with dbmode=3) and for some reason a few phones have multiple entries in the locations table, all with exactly the same 'contact' address. This is causing multiple INVITEs to be sent to the phone, which it doesn't like. Does anyone have an idea how to fix or work around this? Someone advised it may be possible to tell usrloc to only use the most recent contact entry for that phone, can anyone advise how it can be done? You can try this: http://www.mail-archive.com/users@lists.opensips.org/msg09495.html ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] authentication error
On Sun, Aug 1, 2010 at 7:16 PM, Meftah Tayeb tayeb.mef...@gmail.com wrote: hi i used and enabled opensips mysql module and authentication i added user 1000 with pass 1234 user is connecting corectly but evean if i do any username is connecting why? Opensips will not perform authentication unless you say it to do so. But in your cfg file, all auth code (calls to www_authorize, proxy_authorize, www_challenge, proxy_challenge) are disable. br, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Memcache and calculate_ha1 disabled
Hello, I'm reading about using memcache ( http://www.opensips.org/Resources/DocsTutMemcache) to cache passowords. However, what if i need to disable calculate_ha1? When www_authorize/proxy_authorize completes, how could I know which password AVP was used (ha1 or ha1b)? I think I can concatenate the 2 passwords and cache it. Then when I need to authenticate again I could read from cache, split the passwords and try both. But is there a simpler way? br, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Limitations of b2b_entities?
Hi Anca, On Mon, Jul 19, 2010 at 11:58 PM, Anca Vamanu a...@opensips.org wrote: Hi Takeshi, This was something easy to do so I added the support for this - I extended the client_info structure with 3 more fields: to_dname, from_dname ( here you should put the display name) and req_uri for Request URI. The to_uri and from_uri fields must really contain a URI. Check it out. It is OK now. Thanks. While testing, I have found another thing: if I call server_new and a CANCEL is received before I call send_reply(200 OK), this event is not notified to the b2b_notify_t callback function. It would be nice if you would also make your perl/lua code controlling b2b_entities module public. Sure (if i ever managed to get this done: I'm not a seasoned c/perl/lua programmer and opensips source code keeps mystifying me). regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Limitations of b2b_entities?
Hello, I'm playing with the module b2b_entities. I was thinking in writing some perl/lua code to control UA entities provided by this module. However, I am seeing some shortcomings/bugs in it: 1) If I pass \user1\sip:us...@test1.com sip%3aus...@test1.com as from_uri to function client_new, it will be rendered as this in the request: From: user1sip:us...@test1.com sip%3aus...@test1.com ;tag=a34656f511fe5edd5d76196c64f246a7-aba5 (obs: to_uri is free from this problem: to_uri = \user2\sip:us...@192.168.2.126 sip%3aus...@192.168.2.126 - To: user2sip:us...@192.168.2.126 sip%3aus...@192.168.2.126) 2) function client_new doesn't permit to specify the Request-URI. There are situations where Request-URI will have to be set differently from To-Uri (I know at least the case where some PSTN-GWs recognises RURI-Username != To-URI-Username as redirection indication). I just thought to highlight those points, so that they could be improved in the future. regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URI PARSING
On Fri, Jul 9, 2010 at 8:10 PM, Premalatha Kuppan
premala...@ngintech.comwrote:
Please tell me how to search for special character in a string.
{s.select,0,_} this would return the value; instead if i want to search
whether special characters exist or not, which transformation should i use.
Use module regex instead:
http://www.opensips.org/html/docs/modules/1.6.x/regex.html#id228196
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Registration expiration doesn't decrease usrloc:registered_users
Thanks, Bogdan, I tested and it looks fine now. I will do load tests and report any problems. regards, takeshi On Fri, Jul 9, 2010 at 5:17 PM, Bogdan-Andrei Iancu bog...@voice-system.rowrote: FIXED, please update from SVN. Regards, Bogdan Bogdan-Andrei Iancu wrote: Hi, indeed, I was able to reproduce the error - I will work for a fix. Thanks for reporting, Bogdan mayamatakeshi wrote: On Wed, Jul 7, 2010 at 2:29 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi, what db_mode are you using for usrloc? Hi Bogdan, modparam(usrloc, db_mode, 0) mayamatakeshi wrote: Hello, I'm testing with opensips 1.6, latest revision 6985. I'm not sure if I'm doing something wrong but I'm not seeing the value of usrloc:registered_users to decrease when the registration expires. To test this, I have set a low value for expires. Here are the parameters I have changed in the sample opensips.cfg: modparam(registrar, max_expires, 30) modparam(registrar, min_expires, 30) modparam(registrar, default_expires, 30) modparam(alias_db|auth_db|usrloc|uri, use_domain, 1) Then using SIPp I generated a single REGISTER, got the 200 OK and waited 30 seconds. After that, I got this: [r...@centos53-02008]# opensipsctl fifo get_statistics usrloc: usrloc:registered_users = 1 usrloc:location-users = 1 usrloc:location-contacts = 0 usrloc:location-expires = 1 But I was expecting registered_users to be zero, as it happens when I unregister: After registration: [r...@centos53-02008 Basix]# opensipsctl fifo get_statistics usrloc: usrloc:registered_users = 1 usrloc:location-users = 1 usrloc:location-contacts = 1 usrloc:location-expires = 0 After unregistration (request with Expires=0): [r...@centos53-02008 Basix]# opensipsctl fifo get_statistics usrloc: usrloc:registered_users = 0 usrloc:location-users = 0 usrloc:location-contacts = 0 usrloc:location-expires = 0 regards, takeshi ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 20 - 24 September 2010, Frankfurt, Germany www.voice-system.ro http://www.voice-system.ro ___ Users mailing list Users@lists.opensips.org mailto: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 -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 20 - 24 September 2010, Frankfurt, Germany www.voice-system.ro ___ 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] Registration expiration doesn't decrease usrloc:registered_users
On Wed, Jul 7, 2010 at 2:29 AM, Bogdan-Andrei Iancu bog...@voice-system.rowrote: Hi, what db_mode are you using for usrloc? Hi Bogdan, modparam(usrloc, db_mode, 0) mayamatakeshi wrote: Hello, I'm testing with opensips 1.6, latest revision 6985. I'm not sure if I'm doing something wrong but I'm not seeing the value of usrloc:registered_users to decrease when the registration expires. To test this, I have set a low value for expires. Here are the parameters I have changed in the sample opensips.cfg: modparam(registrar, max_expires, 30) modparam(registrar, min_expires, 30) modparam(registrar, default_expires, 30) modparam(alias_db|auth_db|usrloc|uri, use_domain, 1) Then using SIPp I generated a single REGISTER, got the 200 OK and waited 30 seconds. After that, I got this: [r...@centos53-02008]# opensipsctl fifo get_statistics usrloc: usrloc:registered_users = 1 usrloc:location-users = 1 usrloc:location-contacts = 0 usrloc:location-expires = 1 But I was expecting registered_users to be zero, as it happens when I unregister: After registration: [r...@centos53-02008 Basix]# opensipsctl fifo get_statistics usrloc: usrloc:registered_users = 1 usrloc:location-users = 1 usrloc:location-contacts = 1 usrloc:location-expires = 0 After unregistration (request with Expires=0): [r...@centos53-02008 Basix]# opensipsctl fifo get_statistics usrloc: usrloc:registered_users = 0 usrloc:location-users = 0 usrloc:location-contacts = 0 usrloc:location-expires = 0 regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 20 - 24 September 2010, Frankfurt, Germany www.voice-system.ro ___ 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] URI PARSING
On Wed, Jul 7, 2010 at 7:32 PM, Premalatha Kuppan
premala...@ngintech.comwrote:
Hi,
Iam using avp_db_query, but opensips is dead with following error,
During Registration, when the user is registered. I have to parse and store
th value in DB. So, included the avp_db_query in this loop. Is this right
one to do. I assume the value of rU is NULL over here, probably its failing.
Not sure :(
Any insight ?
if (is_method(REGISTER))
{
# authenticate the REGISTER requests (uncomment to enable
auth)
if (!www_authorize(10.140.1.110, subscriber))
{
www_challenge(10.140.1.110, 0);
exit;
}
# if (!db_check_to())
# {
# sl_send_reply(403,Forbidden auth ID);
# exit;
# }
if (!save(location))
sl_reply_error();
$var(x) = $rU ;
avp_db_query(INSERT INTO TESTTABLE (xxx,yyy,zzz)
VALUES('$(var(x){s.select,0,_})','$var(x)({s.select,1,_})','$(var(x){s.select,2,_})');
exit;
}
Jul 7 06:25:16 204548-4 /usr/local/sbin/opensips[17791]:
ERROR:avpops:fixup_db_query_avp: no db url defined to be used by this
function
Jul 7 06:25:16 204548-4 /usr/local/sbin/opensips[17791]:
*ERROR:core:fix_actions:
fixing failed (code=-6) at cfg line 351*
Jul 7 06:25:16 204548-4 /usr/local/sbin/opensips[17791]: ERROR:core:main:
failed to fix configuration with err code -6
Clearly you misplaced the parenthesis:
$(var(x){s.select,0,_})',
'$var(x)({s.select,1,_})',
'$(var(x){s.select,2,_})'
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URI PARSING
On Wed, Jul 7, 2010 at 8:01 PM, Premalatha Kuppan
premala...@ngintech.comwrote:
Thanks. Corrected teh opensips.cfg file
But the problem was due to this. by default DB URL was NULL. Now i have
added the DB URL for avpops.
# default URL
modparam(avpops,db_url,mysql://user:pas...@host/database)
But the value of $rU is NULL..am i placing it in wrong location?
I want the userpart e.g: sip:userp...@domain is the request uri; i need
only userpart . I believe that would be stored in $rU; but tis coming as
NULL. Which variable or where should i place this command for registration
and de-registration.
Any Idea ?
I think you should be able to use $rU directly like this:
$(rU{s.select,0,_})
However, are you sure there is a username in the request URI?
I know at least two sip phone models (SAXA/03.T1.10.9 and CounterPath
eyebeam), that don't send a username in the REGISTER Request URI:
REGISTER sip:test.com SIP/2.0
Via: SIP/2.0/UDP 192.168.11.214:5060;branch=z9hG4bK23ff25886a332c4f
To: 1694sip:1...@test.com sip%3a1...@test.com
From: 1694sip:1...@test.com sip%3a1...@test.com;tag=6fe26be279236cd3
Call-ID: 40025caa51f66e0d06d5f8595f531...@192.168.11.214
CSeq: 1 REGISTER
Max-Forwards: 70
Expires: 0
Contact: *
User-Agent: SAXA/03.T1.10.9
Content-Length: 0
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URI PARSING
On Tue, Jul 6, 2010 at 3:29 PM, Premalatha Kuppan premala...@ngintech.comwrote: Thanks. Can i call shell script from opensips.cfg ? Iam getting error, parse error in config file, Any idea ? http://www.opensips.org/html/docs/modules/1.6.x/exec.html#id226438 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URI PARSING
On Tue, Jul 6, 2010 at 7:43 PM, Premalatha Kuppan premala...@ngintech.comwrote: I have followed this link and used exec_msg(./test.sh '$$SIP_OUSER'); Opensips is running, but i doubt whether it is the correct way to use it. From log file, i can see DBG:exec:w_exec_msg: executing [./test.sh '$SIP_OUSER'] But, script action is not done. My intention is to execute test.sh script and send Username has input to test.sh. How to use it using exec commands ? I have never used module exec myself. I know about it because I read the opensips docs a lot. Try the sample commands. It they work, yours should too. Then experiment passing the full path to the script instead of a relative path. Keep trying, you will get there. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URI PARSING
On Fri, Jul 2, 2010 at 6:44 PM, Premalatha Kuppan premala...@ngintech.comwrote: Hi, Can anyone tell me how to parse SIP URL in Opensips. Iam getting request as Sip:aa_bb_...@opensips.orgsip%3aaa_bb_...@opensips.org. I need to parse aa_bb_ccc as aa,bb,cc and store it in DB. Anyidea or suggestions ? Maybe you can use this transformation on the $rU: http://www.opensips.org/Resources/DocsCoreTran16#toc6 But you will need to call it 3 times, once for each token. If the number of token varies, then I think you could ask module perl to split it to you. br, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Registration expiration doesn't decrease usrloc:registered_users
Hello, I'm testing with opensips 1.6, latest revision 6985. I'm not sure if I'm doing something wrong but I'm not seeing the value of usrloc:registered_users to decrease when the registration expires. To test this, I have set a low value for expires. Here are the parameters I have changed in the sample opensips.cfg: modparam(registrar, max_expires, 30) modparam(registrar, min_expires, 30) modparam(registrar, default_expires, 30) modparam(alias_db|auth_db|usrloc|uri, use_domain, 1) Then using SIPp I generated a single REGISTER, got the 200 OK and waited 30 seconds. After that, I got this: [r...@centos53-02008]# opensipsctl fifo get_statistics usrloc: usrloc:registered_users = 1 usrloc:location-users = 1 usrloc:location-contacts = 0 usrloc:location-expires = 1 But I was expecting registered_users to be zero, as it happens when I unregister: After registration: [r...@centos53-02008 Basix]# opensipsctl fifo get_statistics usrloc: usrloc:registered_users = 1 usrloc:location-users = 1 usrloc:location-contacts = 1 usrloc:location-expires = 0 After unregistration (request with Expires=0): [r...@centos53-02008 Basix]# opensipsctl fifo get_statistics usrloc: usrloc:registered_users = 0 usrloc:location-users = 0 usrloc:location-contacts = 0 usrloc:location-expires = 0 regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Segfault during load test
Hi Bogdan, On Mon, Apr 12, 2010 at 10:27 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Takeshi, Do you still have the core file ? If yes, could you print dest and *dest ? (gdb) print dest $1 = (void **) 0x0 Do you get any previous errors (from the same process) ? like mem errors? Now that you mentioned, I went to the log file and saw lots of this: Apr 3 05:25:19 centos53_02005 /usr/local/sbin/opensips[29732]: ERROR:tm:relay_reply: no more share memory Apr 3 05:25:19 centos53_02005 /usr/local/sbin/opensips[29732]: ERROR:tm:_reply_light: failed to allocate shmem buffer (sorry, I should have looked into this as first thing) Also, can you reproduce this ? I was load testing r. 6605 with 800 Registrations per second (32000 users) and didn't saw this ever happen before (I ran this same load session several times: one of them reached 1 billion registration requests). And actually, in previous tests I was using opensips with just 256 MB of shared memory and not with -m 1024 as above. With r. 6758 so far nothing wrong happened. I will update to 1.6 head and will report if problem happens again, but maybe in the above case, my SIPp instance went crazy in the other end and generated too much traffic. Thanks and regards, takeshi mayamatakeshi wrote: Hello, I was load testing opensips-1.6 r. 6605 and I got a seg fault. I will try updating to most recent revision of 1.6, but I thought someone would like to take a look at this. regards, takeshi Core was generated by `/usr/local/sbin/opensips -m 1024'. Program terminated with signal 11, Segmentation fault. [New process 29728] #0 slot_add (_s=0x2b361ce6ac10, _r=0x2b363326ca20) at hslot.c:151 151 *dest = _r; (gdb) bt #0 slot_add (_s=0x2b361ce6ac10, _r=0x2b363326ca20) at hslot.c:151 #1 0x2b3619b9ac65 in mem_insert_urecord (_d=0x2b361ce69560, _aor=value optimized out, _r=0x7fff919cb740) at udomain.c:648 #2 0x2b3619b9ad42 in insert_urecord (_d=0x2b361ce69560, _aor=0x7fff919cb958, _r=0x7fff919cb740) at udomain.c:781 #3 0x2b3619fb5769 in insert_contacts (_m=0x79b600, _c=0x79c468, _d=0x2b361ce69560, _a=0x7fff919cb958, _sctx=0x7fff919cb950) at save.c:398 #4 0x2b3619fb6592 in save (_m=0x79b600, _d=0x2b361ce69560 , _f=value optimized out, _s=0x0) at save.c:680 #5 0x0040e9e9 in do_action (a=0x78def8, msg=0x79b600) at action.c:967 #6 0x004114bc in run_action_list (a=value optimized out, msg=0x79b600) at action.c:139 #7 0x00466b24 in eval_expr (e=0x78dfd0, msg=0x79b600, val=0x0) at route.c:1240 #8 0x0046659d in eval_expr (e=0x78e020, msg=0x79b600, val=0x0) at route.c:1553 #9 0x00466545 in eval_expr (e=0x78e070, msg=0x79b600, val=0x0) at route.c:1558 #10 0x0040d9cd in do_action (a=0x78e2c8, msg=0x79b600) at action.c:689 #11 0x004114bc in run_action_list (a=value optimized out, msg=0x79b600) at action.c:139 #12 0x0040fcc3 in do_action (a=0x78e888, msg=0x79b600) at action.c:706 #13 0x004114bc in run_action_list (a=value optimized out, msg=0x79b600) at action.c:139 #14 0x00411819 in run_top_route (a=0x78ad18, msg=0x79b600) at action.c:119 #15 0x00455ff5 in receive_msg ( buf=0x754f20 REGISTER sip:domain13564.com http://domain13564.com SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.2.6:;rport;branch=z9hG4bK-4540-1443964-5\r\nFrom: \13564\ sip:13...@domain13564.com sip%3a13...@domain13564.com mailto:sip%3a13...@domain13564.com sip%253a13...@domain13564.com ;tag=4540SIPpTag001443964\r\nTo: \13564\ sip:13564@..., len=575, rcv_info=0x7fff919ccc10) at receive.c:162 #16 0x0049a304 in udp_rcv_loop () at udp_server.c:492 #17 0x00429bbd in main (argc=3, argv=value optimized out) at main.c:818 (gdb) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Segfault during load test
Hello, I was load testing opensips-1.6 r. 6605 and I got a seg fault. I will try updating to most recent revision of 1.6, but I thought someone would like to take a look at this. regards, takeshi Core was generated by `/usr/local/sbin/opensips -m 1024'. Program terminated with signal 11, Segmentation fault. [New process 29728] #0 slot_add (_s=0x2b361ce6ac10, _r=0x2b363326ca20) at hslot.c:151 151 *dest = _r; (gdb) bt #0 slot_add (_s=0x2b361ce6ac10, _r=0x2b363326ca20) at hslot.c:151 #1 0x2b3619b9ac65 in mem_insert_urecord (_d=0x2b361ce69560, _aor=value optimized out, _r=0x7fff919cb740) at udomain.c:648 #2 0x2b3619b9ad42 in insert_urecord (_d=0x2b361ce69560, _aor=0x7fff919cb958, _r=0x7fff919cb740) at udomain.c:781 #3 0x2b3619fb5769 in insert_contacts (_m=0x79b600, _c=0x79c468, _d=0x2b361ce69560, _a=0x7fff919cb958, _sctx=0x7fff919cb950) at save.c:398 #4 0x2b3619fb6592 in save (_m=0x79b600, _d=0x2b361ce69560 , _f=value optimized out, _s=0x0) at save.c:680 #5 0x0040e9e9 in do_action (a=0x78def8, msg=0x79b600) at action.c:967 #6 0x004114bc in run_action_list (a=value optimized out, msg=0x79b600) at action.c:139 #7 0x00466b24 in eval_expr (e=0x78dfd0, msg=0x79b600, val=0x0) at route.c:1240 #8 0x0046659d in eval_expr (e=0x78e020, msg=0x79b600, val=0x0) at route.c:1553 #9 0x00466545 in eval_expr (e=0x78e070, msg=0x79b600, val=0x0) at route.c:1558 #10 0x0040d9cd in do_action (a=0x78e2c8, msg=0x79b600) at action.c:689 #11 0x004114bc in run_action_list (a=value optimized out, msg=0x79b600) at action.c:139 #12 0x0040fcc3 in do_action (a=0x78e888, msg=0x79b600) at action.c:706 #13 0x004114bc in run_action_list (a=value optimized out, msg=0x79b600) at action.c:139 #14 0x00411819 in run_top_route (a=0x78ad18, msg=0x79b600) at action.c:119 #15 0x00455ff5 in receive_msg ( buf=0x754f20 REGISTER sip:domain13564.com SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.2.6:;rport;branch=z9hG4bK-4540-1443964-5\r\nFrom: \13564\ sip:13...@domain13564.com sip%3a13...@domain13564.com;tag=4540SIPpTag001443964\r\nTo: \13564\ sip:13564@..., len=575, rcv_info=0x7fff919ccc10) at receive.c:162 #16 0x0049a304 in udp_rcv_loop () at udp_server.c:492 #17 0x00429bbd in main (argc=3, argv=value optimized out) at main.c:818 (gdb) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Using is_from_gw with incoming-only GWs
While testing with drouting function is_from_gw(), it seems that in addition to be present in the dr_gateways table, there must also exist at least one entry in dr_rules referencing the GW otherwise the function returns false. Is this intentional? In my case, I have incoming-only GWs so there are no rules for them. For now for each of them I add a rule that never matches just to permit to use is_from_gw, but I'm might be overlooking something. regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Using is_from_gw with incoming-only GWs
On Thu, Apr 1, 2010 at 4:28 PM, mayamatakeshi mayamatake...@gmail.comwrote: While testing with drouting function is_from_gw(), it seems that in addition to be present in the dr_gateways table, there must also exist at least one entry in dr_rules referencing the GW otherwise the function returns false. Is this intentional? In my case, I have incoming-only GWs so there are no rules for them. For now for each of them I add a rule that never matches just to permit to use is_from_gw, but I'm might be overlooking something. Sorry, please ignore. Doing some more tests i realized for is_from_gw to work, dr_rules cannot be empty (I was testing with the minimal changes in the db so I have not added the records for the other GWs). regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact list in REGISTER response
On Wed, Feb 24, 2010 at 8:03 PM, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Be careful about the expires value - the usrloc module will choose an
expire value based on the received expire and the max_expire (module
param). You need to send back to the UAC exactly the same expire as the
own determined by usrloc, otherwise the server and client will be out of
sync from expiration point of view. Like the uac will think it is still
registered, will the proxy will expire the UAC's contact.
Ah. Got it.
Thanks, Bogdan.
regards,
takeshi
mayamatakeshi wrote:
On Tue, Feb 16, 2010 at 6:45 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro mailto:bog...@voice-system.ro wrote:
Hi,
Yes, that is the only way - if REGISTRAR module builds the reply, it
will do it in RFC way (adding all the registered contacts) and you
cannot change this behaviour.
Thanks,
I tested it:
if (!save(location,rp1)) {
sl_reply_error();
}
append_to_reply(Contact: $ct;expires=40\r\n);
sl_send_reply(200,OK);
It works fine.
regards,
takeshi
mayamatakeshi wrote:
On Tue, Feb 16, 2010 at 12:54 AM, mayamatakeshi
mayamatake...@gmail.com mailto:mayamatake...@gmail.com
mailto:mayamatake...@gmail.com mailto:mayamatake...@gmail.com
wrote:
Hello,
about registration, is it possible somehow to make opensips to
reply with a 200 OK containing only the Contact of the
registering UA instead of all contacts from usrloc?
I'm having a problem with eyebeam. It has a bug (at least the
version I'm testing) and it doesn't parse the Contact header
correctly and it always gets the value of expires from the
first
contact listed.
So in the case of a 200 OK with a Contact like this:
Contact: sip:1...@192.168.2.5:5050
http://sip:1...@192.168.2.5:5050
http://sip:1...@192.168.2.5:5050;expires=5;received=sip:
192.168.2.5:5050
http://192.168.2.5:5050
http://192.168.2.5:5050, sip:1...@192.168.0.2:8286
http://sip:1...@192.168.0.2:8286
http://sip:1...@192.168.0.2:8286;expires=40;received=sip:
192.168.128.33:61717
http://192.168.128.33:61717
http://192.168.128.33:61717
it should get expires=40 but it is getting expires=5. And since
eyebeam re-registers 5 seconds before expirations, it sends
REGISTER immediately and this goes on in a loop till the
expires
of the first contact gets greater than 5 (when the other
terminal
re-registers).
I have not tested yet, but I think I got it:
I have to call the function save with the flag r (no Reply),
compose
the Contact header myself and send the reply.
___
Users mailing list
Users@lists.opensips.org mailto:Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro http://www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org mailto: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
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
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] Contact list in REGISTER response
On Tue, Feb 16, 2010 at 6:45 PM, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Hi,
Yes, that is the only way - if REGISTRAR module builds the reply, it
will do it in RFC way (adding all the registered contacts) and you
cannot change this behaviour.
Thanks,
I tested it:
if (!save(location,rp1)) {
sl_reply_error();
}
append_to_reply(Contact: $ct;expires=40\r\n);
sl_send_reply(200,OK);
It works fine.
regards,
takeshi
mayamatakeshi wrote:
On Tue, Feb 16, 2010 at 12:54 AM, mayamatakeshi
mayamatake...@gmail.com mailto:mayamatake...@gmail.com wrote:
Hello,
about registration, is it possible somehow to make opensips to
reply with a 200 OK containing only the Contact of the
registering UA instead of all contacts from usrloc?
I'm having a problem with eyebeam. It has a bug (at least the
version I'm testing) and it doesn't parse the Contact header
correctly and it always gets the value of expires from the first
contact listed.
So in the case of a 200 OK with a Contact like this:
Contact: sip:1...@192.168.2.5:5050
http://sip:1...@192.168.2.5:5050;expires=5;received=sip:
192.168.2.5:5050
http://192.168.2.5:5050, sip:1...@192.168.0.2:8286
http://sip:1...@192.168.0.2:8286;expires=40;received=sip:
192.168.128.33:61717
http://192.168.128.33:61717
it should get expires=40 but it is getting expires=5. And since
eyebeam re-registers 5 seconds before expirations, it sends
REGISTER immediately and this goes on in a loop till the expires
of the first contact gets greater than 5 (when the other terminal
re-registers).
I have not tested yet, but I think I got it:
I have to call the function save with the flag r (no Reply), compose
the Contact header myself and send the reply.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Conditional processing of branches after lookup
Hello, is it possible to do conditional processing for each branch after lookup() is called (branch_route) based on data saved with the contact? Specifically, I would like to add a different header like Alert-Info based on the field user_agent. regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Conditional processing of branches after lookup
On Wed, Feb 24, 2010 at 2:28 AM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, is it possible to do conditional processing for each branch after lookup() is called (branch_route) based on data saved with the contact? Specifically, I would like to add a different header like Alert-Info based on the field user_agent. Reading the docs, I believe this cannot be done. But a limited conditional processing could be done indirectly using bflags: we can identify the User-Agent during registration and set bflags based on it. Then those bflags will be available in the branches when lookup() is performed. regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Contact list in REGISTER response
Hello, about registration, is it possible somehow to make opensips to reply with a 200 OK containing only the Contact of the registering UA instead of all contacts from usrloc? I'm having a problem with eyebeam. It has a bug (at least the version I'm testing) and it doesn't parse the Contact header correctly and it always gets the value of expires from the first contact listed. So in the case of a 200 OK with a Contact like this: Contact: sip:1...@192.168.2.5:5050;expires=5;received=sip: 192.168.2.5:5050, sip:1...@192.168.0.2:8286;expires=40;received=sip: 192.168.128.33:61717 it should get expires=40 but it is getting expires=5. And since eyebeam re-registers 5 seconds before expirations, it sends REGISTER immediately and this goes on in a loop till the expires of the first contact gets greater than 5 (when the other terminal re-registers). regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact list in REGISTER response
On Tue, Feb 16, 2010 at 12:54 AM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, about registration, is it possible somehow to make opensips to reply with a 200 OK containing only the Contact of the registering UA instead of all contacts from usrloc? I'm having a problem with eyebeam. It has a bug (at least the version I'm testing) and it doesn't parse the Contact header correctly and it always gets the value of expires from the first contact listed. So in the case of a 200 OK with a Contact like this: Contact: sip:1...@192.168.2.5:5050;expires=5;received=sip: 192.168.2.5:5050, sip:1...@192.168.0.2:8286;expires=40;received=sip: 192.168.128.33:61717 it should get expires=40 but it is getting expires=5. And since eyebeam re-registers 5 seconds before expirations, it sends REGISTER immediately and this goes on in a loop till the expires of the first contact gets greater than 5 (when the other terminal re-registers). I have not tested yet, but I think I got it: I have to call the function save with the flag r (no Reply), compose the Contact header myself and send the reply. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips segfault during pua_dialoginfo tests
Hello Anca, it's done. Thanks. br, takeshi On Wed, Nov 25, 2009 at 1:59 AM, Anca Vamanu a...@opensips.org wrote: Hi Takeshi, Please open a bug report on sourceforge tracker http://sourceforge.net/tracker/?atid=1086410group_id=232389func=browse and specify what version you are using. Also can you please run 'bt full' to see also what are the values of the variables there. Regards, -- Anca Vamanu www.voice-system.ro mayamatakeshi wrote: Hello, is the mailing list the place to report segfaults? I'm doing some manual tests with presence and pua dialoginfo modules. Opensips is crashing very frequently. Last time was this: Log: Nov 25 00:57:27 lab kernel: opensips[10924]: segfault at 0008 rip 2b2fe86c65f9 rsp 7fffc3b581e0 error 4 Backtrace: #0 0x2b2fe86c65f9 in publ_build_hdr (expires=3600, ev=0x0, content_type=0x0, etag=0x2b2fe940c248, extra_headers=0x2b2fe940c335, is_body=0) at send_publish.c:80 80 memcpy(str_hdr-s+ str_hdr-len, ev-name.s, ev-name.len); (gdb) bt #0 0x2b2fe86c65f9 in publ_build_hdr (expires=3600, ev=0x0, content_type=0x0, etag=0x2b2fe940c248, extra_headers=0x2b2fe940c335, is_body=0) at send_publish.c:80 #1 0x2b2fe86c24d1 in update_pua (p=0x2b2fe940c1f8, hash_code=value optimized out) at pua.c:670 #2 0x2b2fe86c2f74 in hashT_clean (ticks=value optimized out, param=value optimized out) at pua.c:625 #3 0x0048bf63 in start_timer_processes () at timer.c:325 #4 0x004297e1 in main (argc=3, argv=0x4) at main.c:867 regards, takeshi ___ 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] Avoiding use of table presentity
Thanks, Anka, I got it. best regards. On Fri, Nov 20, 2009 at 10:14 PM, Anca Vamanu a...@opensips.org wrote: mayamatakeshi wrote: Hello, I'm experimenting with module presence. Is it possible somehow to make presence to NOT write to table presentity? I'm sending PUBLISH messages to opensips and I just want its contents to be sent to the watchers. I don't care to add previous contents nor even to send anything in case of NOTIFY due to SUBSCRIBE. regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users Hi Takeshi, No, it is not possible to avoid using presentity table. The presence module uses it real time for storing info from Publish messages and getting info when sending Notifies. The reason why the database must be used in this case is that the documents can be large and can not be stored in memory. Regards, -- Anca Vamanu www.voice-system.ro ___ 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] NAT handling on a chain of proxies
On Tue, Nov 17, 2009 at 12:00 AM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi Takeshi,
First of all, you want to do the detection for the initial requests or
for sequential requests only?
Hi Bogdan,
I needed to check and change both initial and subsequent requests.
But going to the bottom, you should do the fixing only if the contact
show signs of NAT presence (network level or private ips) - so you
should not rely on the position of your proxy in the chain, but on the
actual information from the messages.
Yes, but in my case, the downstream proxy bridges public and private
networks and so some private ips showing up in the Contact header were valid
and should not be modified and this was causing problems.
So, I decided to check and fix the header Contact only if the source of the
message is not in the GW tables (in this table I put all proxies and GWs I
interconnect with).
So i just have to use is_from_gw (drouting) or from_gw (lcr) and everything
is fine now.
br,
takeshi
mayamatakeshi wrote:
On Fri, Oct 30, 2009 at 7:54 PM, mayamatakeshi
mayamatake...@gmail.com mailto:mayamatake...@gmail.com wrote:
Hello,
is there an easy way to check if my opensips instance is the last
proxy before the UAS?
I need this to avoid calling nat_uac_test/fix_nated_contact in my
onreply_route in case I'm receiving a response relayed by another
proxy.
I was thinking I could test if i'm in the top-most Record-Route in
the reply, but I was unable to figure out how to do it.
Reading the page about transformations, I think something like this
would work:
if ( $(hdr(Record-Route)[0]{uri.host}) == myself ) {
}
However, I think to implement a generic way to handle this would
become very convoluted: I was thinking that I should do this
considering the possibility of dealing with redirection and downstream
proxies that would do not record_route or would do it conditionally.
So I would have to check this, set some parameter in Record-Route,
take care in case of branching and checking the Route header etc, etc.
But I think this is a lot of trouble and will not actually happen in
our network.
So I will write my cfg just testing:
if ( !from_gw() ) {
fix_nated_contact();
}
because I just need to fix the contact of the registered users.
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
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] Presence server behind load-balancer
Hello, I am testing opensips as registrar in the following scenario: UA ---REGISTER--- opensips (load-balancer) REGISTER with header Path opensips (registrar). This works fine, and I have the INVITEs that reach the registrar going thru both proxies as expected: UA -INVITE- opensips(load-balancer) ---INVITE---opensips(registrar) -- GW But now I want to make the registrar to work as a presence server and I want the NOTIFY requests to go thru the load-balancer too (to permit reach UAs behind NAT). Is this possible? I mean, when presence is processed by opensips, it seems to me we cannot route NOTIFY requests using the usrloc data from the registrar, because the NOTIFYs generated by this are not processed by the cfg file. So, is there any way to do this? br, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] NAT handling on a chain of proxies
On Fri, Oct 30, 2009 at 7:54 PM, mayamatakeshi mayamatake...@gmail.comwrote:
Hello,
is there an easy way to check if my opensips instance is the last proxy
before the UAS?
I need this to avoid calling nat_uac_test/fix_nated_contact in my
onreply_route in case I'm receiving a response relayed by another proxy.
I was thinking I could test if i'm in the top-most Record-Route in the
reply, but I was unable to figure out how to do it.
Reading the page about transformations, I think something like this would
work:
if ( $(hdr(Record-Route)[0]{uri.host}) == myself ) {
}
However, I think to implement a generic way to handle this would become very
convoluted: I was thinking that I should do this considering the possibility
of dealing with redirection and downstream proxies that would do not
record_route or would do it conditionally. So I would have to check this,
set some parameter in Record-Route, take care in case of branching and
checking the Route header etc, etc.
But I think this is a lot of trouble and will not actually happen in our
network.
So I will write my cfg just testing:
if ( !from_gw() ) {
fix_nated_contact();
}
because I just need to fix the contact of the registered users.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] NAT handling on a chain of proxies
Hello, is there an easy way to check if my opensips instance is the last proxy before the UAS? I need this to avoid calling nat_uac_test/fix_nated_contact in my onreply_route in case I'm receiving a response relayed by another proxy. I was thinking I could test if i'm in the top-most Record-Route in the reply, but I was unable to figure out how to do it. (to avoid calling fix_nated_contact in requests, i think I can use textops:is_present_hf). regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Module Path and function loose_route
Hello,
I'm testing with the path module.
I'm using this scheme:
User1 - REGISTER - opensips - REGISTER - registrar.
Then, when a call comes into the registrar to User1 I'm trying to send
the INVITE to User1 via opensips.
In my cfg I have added a test to see if an out-of-dialog request
contains a valid route set:
#if there is a pre-loaded route set, then it is a request to
be sent to a user
if(loose_route()) {
...relay the request
}
However, loose_route() is returning false.
This is in accordance to
http://www.opensips.org/html/docs/modules/1.5.x/rr.html where we can
read:
There is only one exception: If the request is out-of-dialog (no
to-tag) and there is only one Route: header indicating the local
proxy, then the Route: header is removed and the function returns
FALSE.
But why does it return FALSE?
I'm trying with the following INVITE:
INVITE sip:eywhywhdyhdyw...@192.168.4.6 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.121:5070;rport;branch=z9hG4bKPjJ5ldIJPcArMnJ0soPgyRWi3zGyA9TYBt
Max-Forwards: 70
From: sip:p...@ldilsjeodlskekdksl.com;tag=QumeNtxM4Cb9Qj77vumy5ujr8vHoJBfS
To: sip:eywhywhdyhdyw...@ldilsjeodlskekdksl.com
Contact: sip:tes...@192.168.2.121:5070
Call-ID: Vw3Cvf6utvMuzj-K2WGOc.YYvYWfORGB
CSeq: 28223 INVITE
Route: sip:192.168.2.100:5060;lr
Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK
Supported: replaces, 100rel
Content-Type: application/sdp
Content-Length: 453
[SDP ommited]
So, how should the INVITE be sent for loose_route to succeed?
Or should I put something else in my cfg?
regards,
takeshi
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Module Path and function loose_route
On Fri, Aug 21, 2009 at 3:43 AM, Iñaki Baz Castilloi...@aliax.net wrote: El Jueves, 20 de Agosto de 2009, Alex Balashov escribió: Iñaki Baz Castillo wrote: El Jueves, 20 de Agosto de 2009, mayamatakeshi escribió: There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE. But why does it return FALSE? Because if an initial request (no To-tag) has a single Route header pointing to the proxy handling it, it's useless. That's correct - initial INVITEs (and all initial requests) are different than in-dialog requests (requests arising within a dialog created by the initial requests). They are routed manually, not using loose_route() in any way. In fact, in case of PATH usage, the registrar should receive the request for a registered user, add Route header pointing to the inbound/outbound proxy of the registered user and change the RURI with the real location of the registered user (or mapped public address in case of NAT), route the request to it, Iñaki, Alex, thanks. But I'm still confused about this. The above is what I believe I'm doing in my tests: I am setting the RURI to the real location of the registered user and I'm passing the URI of opensips in the sole Route header. and the inbound/outbound proxy should remove the Route header and route the request based on the RURI as usual. How my cfg should accomplish this then? If loose_route returned TRUE, I would simply relay the request to the address in the RURI. Do you mean I should not be using loose_route() for this? Should I perform the checking in another way? Well, of course I could come up with something else, but it seems to me loose_route() would be the simplest way to do it if there was not that exception clause that I really don't understand why is there. regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem using OpenSIPS with SIPp for testing
On Sat, Mar 14, 2009 at 2:49 PM, SAMEER KUMAR sameer.kasi2...@gmail.com wrote:
Hi Takeshi,
I used the exact cfg file you sent. But that too seems to be not helping me.
Sameer,
the cfg does work. I tested using an eyebeam phone. The call goes thru
opensips and it is relayed to another terminal.
I debugged to find out that the message processing does not go to the
if(method=INVITE conditional branch.
If the processing of the request doesn't reach that line, then maybe
the message received (generated by SIPp), is not properly formed. Have
you checked the contents of the message?
Also, have you tried to call using some UA like x-lite?
Put some xlog in the cfg and check where and why the above line is not
hit by the processing.
I then added all the t_relay and
other functions in the route[1] branch (anyways relaying would be needed for
ACKs revieved as well). But that too does not seem to be working. I found
the t_relay returns following error in route[1].
Mar 14 16:58:54 [3891] ERROR:tm:t_forward_nonack: no branch for forwarding
Mar 14 16:58:54 [3891] ERROR:tm:w_t_relay: t_forward_nonack failed
Mar 14 16:59:24 [3892] CRITICAL:tm:t_should_relay_response: pick_branch
failed (lowest==-1) for code 408
I am not getting where the things are goinf wrong...
Can you please throw some light.
Sameer Kumar
B.Tech, Computer Engineering
Institute of Technology,
Nirma University,
Ahmedabad, Gujarat
On Fri, Mar 13, 2009 at 8:51 AM, mayamatakeshi mayamatake...@gmail.com
wrote:
On Fri, Mar 13, 2009 at 9:04 PM, SAMEER KUMAR sameer.kasi2...@gmail.com
wrote:
Dear,
Thanks for such a prompt reply.
Earlier I was not even using a saperate processing for INVITE. I was
supposed to be processed by the fefault call to route[1] (please refer
to
the cfg file). But as I noticed the problem I stated, I tought that the
call
to t_relay in route[1] is not being processed (in route[1] in the if
condition check). So I made a explicit processing for INVITE and then
used
t_relay. But when that too did not seemed to help I used stateless
forward.
I had tried lot many different things (like adding via and record route
functions etc etc) and also tested other example cfg files. So be sure
that
there is no problem with cfg file. The problem persists ireespective of
all
that.
Hello Sameer,
my current installation is not compiled with tls and I also don't have
the load_sig_api required by the registrar module. So I cleared your
cfg from registrar functions and running it I can see it does work
(i'm using opensips head r. 5438). Here's the modified cfg file:
debug=3
fork=no
log_stderror=yes
listen=192.168.2.3
port=5080
children=4
dns=yes
rev_dns=yes
#fifo= /tmp/ser_fifo
mpath=/usr/local/lib/opensips/modules
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
modparam(usrloc, db_mode, 0)
modparam(rr, enable_full_lr, 1)
route {
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483, Too Many Hops);
return;
};
if (msg:len max_len) {
sl_send_reply(513, Message Overflow);
return;
};
if (method!=REGISTER) {
record_route();
};
if (loose_route()) {
route(1);
return;
};
if (uri!=myself) {
route(1);
return;
};
if (method==ACK) {
route(1);
return;
}
if (method==INVITE) {
add_local_rport();
force_rport();
forward();
t_relay();
return;
};
route(1);
}
route[1] {
if (!t_relay()) {
sl_reply_error();
};
}
Maybe this is something related to TLS. Have you tried testing with no
TLS?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Profile feature not working
Hello,
I just got the latest svn revision 5330.
I'm trying to use the profile feature of the dialog module to limit the
number of simultaneous calls for a given subscriber.
I got these errors:
Feb 19 01:28:37 ipx029 /usr/local/sbin/opensips[23050]:
ERROR:dialog:set_dlg_profile: dialog was not yet created - script error
Feb 19 01:28:37 ipx029 /usr/local/sbin/opensips[23050]:
ERROR:dialog:w_set_dlg_profile: failed to set profile
Significant portions of the cfg are these:
setflag(4); #set dialog
get_profile_size(caller, $fu, $var(SIZE));
if( $var(SIZE) = $avp(i:10) ){
sl_send_reply(503, Simultaneous calls limit reached);
exit;
}
consume_credentials();
set_dlg_profile(caller,$fu);
This may be a bug as the script was originally written for kamailio 1.4 and
works without problems there.
(I'm aware the dialog module is diverging between the projects, but reading
the docs I could not spot a significant change in this particular feature).
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
