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<body class='hmmessage'><div dir='ltr'>Hi Jeff,<br><br>I would like to have a media server but I don't have one at this moment.<br>Thank you for the response,<br><br>Frank<br><br><div>> From: users-request@lists.opensips.org<br>> Subject: Users Digest, Vol 69, Issue 32<br>> To: users@lists.opensips.org<br>> Date: Sat, 12 Apr 2014 06:39:42 +0200<br>> <br>> Send Users mailing list submissions to<br>> users@lists.opensips.org<br>> <br>> To subscribe or unsubscribe via the World Wide Web, visit<br>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br>> or, via email, send a message with subject or body 'help' to<br>> users-request@lists.opensips.org<br>> <br>> You can reach the person managing the list at<br>> users-owner@lists.opensips.org<br>> <br>> When replying, please edit your Subject line so it is more specific<br>> than "Re: Contents of Users digest..."<br>> <br>> <br>> Today's Topics:<br>> <br>> 1. Re: Miliseconds precision for accounting module (Maciej Bylica)<br>> 2. playing a prompt when backend is down (frank fox)<br>> 3. help playing a prompt when backend is down (frank fox)<br>> 4. Re: help playing a prompt when backend is down (Jeff Pyle)<br>> 5. Issue in VIA Header (B. Buitenhuis)<br>> 6. Re: Miliseconds precision for accounting module (Ryan Mitchell)<br>> <br>> <br>> ----------------------------------------------------------------------<br>> <br>> Message: 1<br>> Date: Fri, 11 Apr 2014 19:06:51 +0200<br>> From: Maciej Bylica <mbsip@gazeta.pl><br>> Subject: Re: [OpenSIPS-Users] Miliseconds precision for accounting<br>> module<br>> To: OpenSIPS users mailling list <users@lists.opensips.org><br>> Message-ID:<br>> <CACynCY7TYA+hHToS7c-oWNBx0m=W_k4sFLfcPXBEOVU4Y3ROEQ@mail.gmail.com><br>> Content-Type: text/plain; charset="iso-8859-1"<br>> <br>> Ryan,<br>> <br>> One more question.<br>> Currently i have some db extra attrs setup. My acc table looks like<br>> following:<br>> <br>> +------------+------------------+------+-----+---------+----------------+<br>> <br>> | Field | Type | Null | Key | Default | Extra |<br>> <br>> +------------+------------------+------+-----+---------+----------------+<br>> <br>> | id | int(10) unsigned | NO | PRI | NULL | auto_increment |<br>> <br>> | method | char(16) | NO | | | |<br>> <br>> | from_tag | char(64) | NO | | | |<br>> <br>> | to_tag | char(64) | NO | | | |<br>> <br>> | callid | char(64) | NO | MUL | | |<br>> <br>> | sip_code | char(3) | NO | | | |<br>> <br>> | sip_reason | char(32) | NO | | | |<br>> <br>> | time | datetime | NO | | NULL | |<br>> <br>> | duration | int(11) unsigned | NO | | 0 | |<br>> <br>> | setuptime | int(11) unsigned | NO | | 0 | |<br>> <br>> | SourceAddr | char(30) | NO | | NULL | |<br>> <br>> | DestAddr | char(30) | NO | | NULL | |<br>> <br>> | Anum | char(30) | NO | | NULL | |<br>> <br>> | Bnum_rU | char(30) | NO | | NULL | |<br>> <br>> | Bnum_tU | char(30) | NO | | NULL | |<br>> <br>> | created | datetime | YES | | NULL | |<br>> <br>> +------------+------------------+------+-----+---------+----------------+<br>> <br>> <br>> modparam("acc", "db_extra", "SourceAddr=$si; DestAddr=$rd; Anum=$fU;<br>> Bnum_rU=$rU; Bnum_tU=$tU")<br>> <br>> <br>> Now using additional data like $time will give me the exact moment the call<br>> is ended, nothing more, am i right?<br>> <br>> To have detailed call duration i need to know exact answer and disconnect<br>> timestamps.<br>> <br>> <br>> Btw: i am using OpenSIPS (1.9.1-notls (x86_64/linux))<br>> <br>> <br>> Thanks,<br>> <br>> Mac<br>> <br>> <br>> 2014-04-10 22:03 GMT+02:00 Ryan Mitchell <rjm@tcl.net>:<br>> <br>> > Using db_extra to stuff custom data into your acc table, use the $time var<br>> > with a format such as "%s.%N" or similar.<br>> ><br>> > Or, as you suggested, do it on the database level with a trigger or<br>> > auto-update column.<br>> ><br>> ><br>> ><br>> > On Thu, Apr 10, 2014 at 10:01 AM, Maciej Bylica <mbsip@gazeta.pl> wrote:<br>> ><br>> >> Hello<br>> >><br>> >> I just want to know how to achieve miliseconds precision for accounting<br>> >> module.<br>> >> This is quite important while trying to sum up total traffic duration<br>> >> with the accuracy of hundred of ms.<br>> >><br>> >> As i see there is no rounding feature implemented as well, but heaving ms<br>> >> precision it could be done directly on DB level.<br>> >><br>> >> Could somebody give me a hand.<br>> >><br>> >> Thanks in advanced<br>> >> Mac<br>> >><br>> >><br>> >><br>> >> _______________________________________________<br>> >> Users mailing list<br>> >> Users@lists.opensips.org<br>> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br>> >><br>> >><br>> ><br>> ><br>> > --<br>> > Ryan Mitchell <rjm@tcl.net><br>> > Telecom Logic, LLC<br>> ><br>> ><br>> > _______________________________________________<br>> > Users mailing list<br>> > Users@lists.opensips.org<br>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br>> ><br>> ><br>> -------------- next part --------------<br>> An HTML attachment was scrubbed...<br>> URL: <http://lists.opensips.org/pipermail/users/attachments/20140411/1ab46234/attachment-0001.htm><br>> <br>> ------------------------------<br>> <br>> Message: 2<br>> Date: Fri, 11 Apr 2014 17:51:36 -0400<br>> From: frank fox <franklyfox@outlook.com><br>> Subject: [OpenSIPS-Users] playing a prompt when backend is down<br>> To: "users@lists.opensips.org" <users@lists.opensips.org><br>> Message-ID: <BAY177-W4412F902322E9B93A6598BC8540@phx.gbl><br>> Content-Type: text/plain; charset="iso-8859-1"<br>> <br>> Greetings,<br>> <br>> Is there any recommendations for my proxy to play an audio through rtpproxy when there is no response sent back from my UAS? I would like to play a notify prompt to inform users that my system is down.<br>> <br>> Can one of these two functions really work for my situation or there are something else?<br>> rtpproxy_stream2uac<br>> minor_branch_flag<br>> <br>> Thank you,<br>> Frank<br>> <br>> -------------- next part --------------<br>> An HTML attachment was scrubbed...<br>> URL: <http://lists.opensips.org/pipermail/users/attachments/20140411/004cf9be/attachment-0001.htm><br>> <br>> ------------------------------<br>> <br>> Message: 3<br>> Date: Fri, 11 Apr 2014 18:00:06 -0400<br>> From: frank fox <franklyfox@outlook.com><br>> Subject: [OpenSIPS-Users] help playing a prompt when backend is down<br>> To: "users@lists.opensips.org" <users@lists.opensips.org><br>> Message-ID: <BAY177-W14EEEFAB87E568D2EC1B97C8540@phx.gbl><br>> Content-Type: text/plain; charset="iso-8859-1"<br>> <br>> <br>> <br>> <br>> <br>> Greetings,<br>> <br>> Is there any recommendations for my proxy to play an audio through rtpproxy when there is no response sent back from my UAS? I would like to play a notify prompt to inform users that my system is down.<br>> <br>> Can one of these two functions really work for my situation or there are something else?<br>> rtpproxy_stream2uac<br>> minor_branch_flag<br>> <br>> Thank you,<br>> Frank<br>> <br>> -------------- next part --------------<br>> An HTML attachment was scrubbed...<br>> URL: <http://lists.opensips.org/pipermail/users/attachments/20140411/c13c93cc/attachment-0001.htm><br>> <br>> ------------------------------<br>> <br>> Message: 4<br>> Date: Fri, 11 Apr 2014 18:02:33 -0400<br>> From: Jeff Pyle <jpyle@fidelityvoice.com><br>> Subject: Re: [OpenSIPS-Users] help playing a prompt when backend is<br>> down<br>> To: OpenSIPS users mailling list <users@lists.opensips.org><br>> Message-ID:<br>> <CACYJG3KV9qnNzz=yWv8S-mDfCiq24RE39WF6GSPCZxV8WGCh8g@mail.gmail.com><br>> Content-Type: text/plain; charset="iso-8859-1"<br>> <br>> I catch the 408 in a failure_route and re-route (t_relay) the call to a<br>> media server to play the recording. I haven't tried any rtpproxy-specific<br>> methods.<br>> <br>> <br>> - Jeff<br>> <br>> <br>> On Fri, Apr 11, 2014 at 6:00 PM, frank fox <franklyfox@outlook.com> wrote:<br>> <br>> ><br>> > Greetings,<br>> ><br>> > Is there any recommendations for my proxy to play an audio through<br>> > rtpproxy when there is no response sent back from my UAS? I would like to<br>> > play a notify prompt to inform users that my system is down.<br>> ><br>> > Can one of these two functions really work for my situation or there are<br>> > something else?<br>> > rtpproxy_stream2uac<br>> > minor_branch_flag<br>> ><br>> > Thank you,<br>> > Frank<br>> ><br>> > _______________________________________________<br>> > Users mailing list<br>> > Users@lists.opensips.org<br>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br>> ><br>> ><br>> -------------- next part --------------<br>> An HTML attachment was scrubbed...<br>> URL: <http://lists.opensips.org/pipermail/users/attachments/20140411/a78ed599/attachment-0001.htm><br>> <br>> ------------------------------<br>> <br>> Message: 5<br>> Date: Fri, 11 Apr 2014 22:09:49 +0000<br>> From: "B. Buitenhuis" <bernard@buitenhuis.nu><br>> Subject: [OpenSIPS-Users] Issue in VIA Header<br>> To: "users@lists.opensips.org" <users@lists.opensips.org><br>> Message-ID: <CF6E34CD.17B1F%bernard@buitenhuis.nu><br>> Content-Type: text/plain; charset="us-ascii"<br>> <br>> Hello,<br>> <br>> I'm facing the same problem as:<br>> https://github.com/OpenSIPS/opensips/issues/190<br>> <br>> When I look at a wireshark, I see the following in the VIA header:<br>> Via: SIP/2.0/UDP \000\000\000\000\000\000\000\000\000\000\000\000\000:5060;branch=z9hG4bK9414.22d66c76.0<br>> <br>> As far as I can see it's only with the CANCEL method, in the config it looks like:<br>> <br>> if (method=="CANCEL")<br>> {<br>> set_advertised_address(185.XX.XX.XX);<br>> <br>> route("PROCESS_RTP_PROXY");<br>> unforce_rtp_proxy();<br>> if (t_check_trans())<br>> t_relay();<br>> exit;<br>> }<br>> <br>> <br>> I've tested it with 1.8.3 and 1.10.1<br>> <br>> Does anyone has this same problem?<br>> <br>> <br>> Regards,<br>> <br>> Bernard Buitenhuis<br>> <br>> -------------- next part --------------<br>> An HTML attachment was scrubbed...<br>> URL: <http://lists.opensips.org/pipermail/users/attachments/20140411/f7519d71/attachment-0001.htm><br>> <br>> ------------------------------<br>> <br>> Message: 6<br>> Date: Fri, 11 Apr 2014 21:39:33 -0700<br>> From: Ryan Mitchell <rjm@tcl.net><br>> Subject: Re: [OpenSIPS-Users] Miliseconds precision for accounting<br>> module<br>> To: OpenSIPS users mailling list <users@lists.opensips.org><br>> Message-ID:<br>> <CAMnAngA25bNqyjM=ROnMO3Y4KcCKDs083ez+iuQoj2Ds-_iKTA@mail.gmail.com><br>> Content-Type: text/plain; charset="iso-8859-1"<br>> <br>> Hello Mac,<br>> <br>> Each row in the acc table is for a transaction. To make a proper CDR out<br>> of the data, you have to combine rows to find the start and end of the<br>> call. That can be harder than it sounds, especially with forking<br>> (parallel, or the more common case of serial forking when you are LCR<br>> routing or simply sending calls to alt destinations after a timeout). I<br>> wrote scripts that implement a simple dialog state machine to make sense of<br>> all the distinct legs of a call, though there should be an easier way with<br>> the auto-cdr / multi call-legs accounting feature of the acc module (anyone<br>> comment on this please?).<br>> <br>> The time field in the acc table will be the timestamp of the response for<br>> the given transaction. If you assign an extra field for another timestamp,<br>> it will depend on where you assign that var in your script. In my case I<br>> assign it in the main routing section so the timestamp indicates the start<br>> of the transaction.<br>> <br>> best regards,<br>> Ryan<br>> <br>> <br>> <br>> On Fri, Apr 11, 2014 at 10:06 AM, Maciej Bylica <mbsip@gazeta.pl> wrote:<br>> <br>> > Ryan,<br>> ><br>> > One more question.<br>> > Currently i have some db extra attrs setup. My acc table looks like<br>> > following:<br>> ><br>> > +------------+------------------+------+-----+---------+----------------+<br>> ><br>> > | Field | Type | Null | Key | Default | Extra |<br>> ><br>> > +------------+------------------+------+-----+---------+----------------+<br>> ><br>> > | id | int(10) unsigned | NO | PRI | NULL | auto_increment |<br>> ><br>> > | method | char(16) | NO | | | |<br>> ><br>> > | from_tag | char(64) | NO | | | |<br>> ><br>> > | to_tag | char(64) | NO | | | |<br>> ><br>> > | callid | char(64) | NO | MUL | | |<br>> ><br>> > | sip_code | char(3) | NO | | | |<br>> ><br>> > | sip_reason | char(32) | NO | | | |<br>> ><br>> > | time | datetime | NO | | NULL | |<br>> ><br>> > | duration | int(11) unsigned | NO | | 0 | |<br>> ><br>> > | setuptime | int(11) unsigned | NO | | 0 | |<br>> ><br>> > | SourceAddr | char(30) | NO | | NULL | |<br>> ><br>> > | DestAddr | char(30) | NO | | NULL | |<br>> ><br>> > | Anum | char(30) | NO | | NULL | |<br>> ><br>> > | Bnum_rU | char(30) | NO | | NULL | |<br>> ><br>> > | Bnum_tU | char(30) | NO | | NULL | |<br>> ><br>> > | created | datetime | YES | | NULL | |<br>> ><br>> > +------------+------------------+------+-----+---------+----------------+<br>> ><br>> ><br>> > modparam("acc", "db_extra", "SourceAddr=$si; DestAddr=$rd; Anum=$fU;<br>> > Bnum_rU=$rU; Bnum_tU=$tU")<br>> ><br>> ><br>> > Now using additional data like $time will give me the exact moment the<br>> > call is ended, nothing more, am i right?<br>> ><br>> > To have detailed call duration i need to know exact answer and disconnect<br>> > timestamps.<br>> ><br>> ><br>> > Btw: i am using OpenSIPS (1.9.1-notls (x86_64/linux))<br>> ><br>> ><br>> > Thanks,<br>> ><br>> > Mac<br>> ><br>> ><br>> > 2014-04-10 22:03 GMT+02:00 Ryan Mitchell <rjm@tcl.net>:<br>> ><br>> >> Using db_extra to stuff custom data into your acc table, use the $time<br>> >> var with a format such as "%s.%N" or similar.<br>> >><br>> >> Or, as you suggested, do it on the database level with a trigger or<br>> >> auto-update column.<br>> >><br>> >><br>> >><br>> >> On Thu, Apr 10, 2014 at 10:01 AM, Maciej Bylica <mbsip@gazeta.pl> wrote:<br>> >><br>> >>> Hello<br>> >>><br>> >>> I just want to know how to achieve miliseconds precision for accounting<br>> >>> module.<br>> >>> This is quite important while trying to sum up total traffic duration<br>> >>> with the accuracy of hundred of ms.<br>> >>><br>> >>> As i see there is no rounding feature implemented as well, but heaving<br>> >>> ms precision it could be done directly on DB level.<br>> >>><br>> >>> Could somebody give me a hand.<br>> >>><br>> >>> Thanks in advanced<br>> >>> Mac<br>> >>><br>> >>><br>> >>><br>> >>> _______________________________________________<br>> >>> Users mailing list<br>> >>> Users@lists.opensips.org<br>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br>> >>><br>> >>><br>> >><br>> >><br>> >> --<br>> >> Ryan Mitchell <rjm@tcl.net><br>> >> Telecom Logic, LLC<br>> >><br>> >><br>> >> _______________________________________________<br>> >> Users mailing list<br>> >> Users@lists.opensips.org<br>> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br>> >><br>> >><br>> ><br>> > _______________________________________________<br>> > Users mailing list<br>> > Users@lists.opensips.org<br>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br>> ><br>> ><br>> <br>> <br>> -- <br>> Ryan Mitchell <rjm@tcl.net><br>> Telecom Logic, LLC<br>> -------------- next part --------------<br>> An HTML attachment was scrubbed...<br>> URL: <http://lists.opensips.org/pipermail/users/attachments/20140411/f14c1838/attachment.htm><br>> <br>> ------------------------------<br>> <br>> _______________________________________________<br>> Users mailing list<br>> Users@lists.opensips.org<br>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br>> <br>> <br>> End of Users Digest, Vol 69, Issue 32<br>> *************************************<br></div> </div></body>
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