Hi, Opensips 1.9 stops working after a successful restart. I tried changing the debug level but all I can get is this below Error in log.
Pls help.. Jan 25 02:56:23 debiansip01 /sbin/opensips[5967]: CRITICAL:db_mysql:wrapper_single_mysql_real_query: driver error (1054): Unknown column 'attr' in 'field list' Jan 25 02:56:23 debiansip01 /sbin/opensips[5967]: ERROR:core:db_do_query: error while submitting query - [select username,contact,expires,q,callid,cseq,flags,cflags,user_agent,received,path ,socket,methods,last_modified,sip_instance,attr from aliases ] Jan 25 02:56:23 debiansip01 /sbin/opensips[5967]: ERROR:usrloc:preload_udomain: db_query (1) failed Jan 25 02:56:23 debiansip01 /sbin/opensips[5967]: ERROR:usrloc:child_init: child(1): failed to preload domain 'aliases' Jan 25 02:56:23 debiansip01 /sbin/opensips[5967]: ERROR:core:init_mod_child: failed to initializing module usrloc, rank 1 Jan 25 02:56:23 debiansip01 /sbin/opensips[5967]: ERROR:core:main_loop: init_child failed for UDP listener Thanks -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of [email protected] Sent: Saturday, February 1, 2014 5:57 AM To: [email protected] Subject: Users Digest, Vol 67, Issue 1 Send Users mailing list submissions to [email protected] To subscribe or unsubscribe via the World Wide Web, visit http://lists.opensips.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to [email protected] You can reach the person managing the list at [email protected] When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: CDRTool - Rating Origination and Termination differently (David M. Lee) 2. topology hiding (BJ Quinn) ---------------------------------------------------------------------- Message: 1 Date: Fri, 31 Jan 2014 11:03:51 -0600 From: "David M. Lee" <[email protected]> Subject: Re: [OpenSIPS-Users] CDRTool - Rating Origination and Termination differently To: OpenSIPS users mailling list <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset="windows-1252" Sorry for the delayed response. I?ll take a look and see if I can correct the patch. Thanks! -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org On Jan 28, 2014, at 6:12 AM, Tijmen de Mes <[email protected]> wrote: > Hi, > > I run into trouble and reversed the patch. In the rating engine (telnet) DebitBalance function does not return anything anymore with the patch applied, so something goes wrong. > > -- > Tijmen de Mes > AG-Projects > > On 27 januari 2014 at 17:55:01, Tijmen de Mes ([email protected]) wrote: > >> Hi >> >> I tested the patch and for now I don?t see any problem with it. Before adding it to trunk, can you update the rating docs and record all changes in 1 patch using darcs? After that I can apply it directly on trunk with your credentials. >> >> These instructions to submit the patch also apply on CDRTool: >> >> http://sipsimpleclient.org/projects/sipsimpleclient/wiki/SipSupport >> >> You don?t need to open a ticket however, just tell me after you send the patch. >> >> -- >> Tijmen de Mes >> AG-Projects >> >> On 26 januari 2014 at 02:08:50, Duane Larson ([email protected]) wrote: >> >>> Ugh.... Nevermind. My "Max Duration" was set to 6 on the Destinations rate setup. I set it to zero and I think things are looking better now. Think I'm done for the day. Good work David! Really appreciate the patch. >>> >>> >>> On Sat, Jan 25, 2014 at 6:56 PM, Duane Larson <[email protected]> wrote: >>> Not sure my "price" is being calculated correctly. >>> >>> My audio.outbound should be $0.005 >>> My audio.inbound should be $0.0035 >>> >>> Here is an example of a call that lasted 2:06 minutes and is >>> audio.outbound >>> Increment=6 MinDuration=6 MaxDuration=6 ConnectFee=0.0000 >>> CallId=dc1e32b1ba18ff91a3 >>> 67aa8df81e8e3c@0:0:0:0:0:0:0:0 Span=1 Duration=6 DestId=1 default >>> Profile=USA_Default Period=weekend Rate=USA_Default Interval=0-24 >>> Cos >>> t=0.0050/60 Price=0.0005 PriceIn=0.0000 >>> >>> Here is an example of a call that lasted 1:33 minutes and is >>> audio.inbound >>> Increment=6 MinDuration=6 MaxDuration=6 ConnectFee=0.0000 >>> CallId=2e0334513a34964e7c0 >>> [email protected]:5060 Span=1 Duration=6 DestId=1 default >>> Profile=USA_Default Period=weekend Rate=USA_Default Interval=0-24 C >>> ost=0.0035/60 Price=0.0003 PriceIn=0.0000 >>> >>> Looks like no matter what the duration is for the inbound calls it costs $0.0003 and the outbound calls all cost $0.0005. >>> >>> >>> >>> >>> On Sat, Jan 25, 2014 at 6:24 PM, Duane Larson <[email protected]> wrote: >>> David/Tijmen/Adrian, >>> >>> It is working for me too. Both inbound and outbound are being recognized and the different rates are being applied. I will keep looking at it while more calls are being made to make sure there are no unforeseen issues. >>> >>> >>> On Fri, Jan 24, 2014 at 5:35 AM, Tijmen de Mes <[email protected]> wrote: >>> Hi David, >>> >>> Thanks for the patch. If have some time Monday I will analyze it and give you some feedback so we include this in CDRTool. >>> >>> Besides the rating, I?ve to check if the code that now sets the the ?route? for the CDRs and if there are no problems. >>> >>> >>> -- >>> Tijmen de Mes >>> AG-Projects >>> >>> On 23 januari 2014 at 21:21:26, David M. Lee ([email protected]) wrote: >>> >>>> I?ve got a first attempt at a patch: >>>> https://gist.github.com/leedm777/8585690 >>>> >>>> To get this to work, your CDR?s will have to record the application >>>> subtype. For me, I set $avp(s:call_class) = ?audio.inbound? or >>>> ?audio.outbound? as appropriate in the routes, and added >>>> Sip-Application-Type=$avp(s:call_class) to ?radius_extra?. >>>> >>>> Billing rates will be matched on the full application, allowing >>>> different rates depending on the application subtype. >>>> >>>> It works with the simple testing I?ve been able to do on my desktop. >>>> Feedback, of course, is greatly appreciated. >>>> >>>> Duane - Does the patch work for you? >>>> >>>> Adrian - Any objections/concerns with this approach? >>>> >>>> If it looks good, I?ll work on updating the docs. >>>> >>>> Thanks! >>>> -- >>>> David M. Lee >>>> Digium, Inc. | Software Developer >>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out >>>> at: www.digium.com & www.asterisk.org >>>> >>>> On Jan 23, 2014, at 4:43 AM, Adrian Georgescu <[email protected]> wrote: >>>> >>>> > I think it would be a good idea. >>>> > >>>> > Adrian >>>> > >>>> > On 22 Jan 2014, at 16:58, David Lee (digium) <[email protected]> wrote: >>>> > >>>> >> Duane Larson wrote: >>>> >>> I have been playing with CDRTool for a while but I am not sure >>>> >>> if it is possible to rate Origination (Inbound) calls >>>> >>> differently than Termination >>>> >>> (Outbound) calls from my SIP Provider. For Origination I pay >>>> >>> 0.0035 and for Termination I pay 0.005. Keep in mind these >>>> >>> costs are for destination "1?. >>>> >> >>>> >> I've recently run into nearly the same situation. >>>> >> >>>> >> I tried overloading the Sip-Application-Type field, but it's >>>> >> limited by the supportedApplicationTypes array in >>>> >> cdr_generic.php. There also seems to be hard coded logic for the different application types. >>>> >> >>>> >> I think I will patch CDRTool so that you can have subtypes of >>>> >> application types (audio.inbound, audio.outbound, etc.). This >>>> >> would allow the billing rates to be a bit more specific for >>>> >> these situations. >>>> >> >>>> >> Thoughts? >>>> >> -- >>>> >> David M. Lee >>>> >> Digium, Inc. | Software Developer >>>> >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out >>>> >> at: www.digium.com & www.asterisk.org >>>> >> _______________________________________________ >>>> >> Users mailing list >>>> >> [email protected] >>>> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> > >>>> > _______________________________________________ >>>> > Users mailing list >>>> > [email protected] >>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> [email protected] >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> -- >>> -- >>> *--*--*--*--*--* >>> Duane >>> *--*--*--*--*--* >>> -- >>> >>> >>> >>> -- >>> -- >>> *--*--*--*--*--* >>> Duane >>> *--*--*--*--*--* >>> -- >>> >>> >>> >>> -- >>> -- >>> *--*--*--*--*--* >>> Duane >>> *--*--*--*--*--* >>> -- >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.opensips.org/pipermail/users/attachments/20140131/9b371062/att achment-0001.htm> ------------------------------ Message: 2 Date: Fri, 31 Jan 2014 18:26:53 -0600 (CST) From: BJ Quinn <[email protected]> Subject: [OpenSIPS-Users] topology hiding To: [email protected] Message-ID: <[email protected]> Content-Type: text/plain; charset=utf-8 Hi, I'd like to use topology_hiding(), but I can't quite understand how to integrate it into the routing part of the configuration file. I have my opensips box on a public IP and some machines initiating calls through the opensips box that are also on public IPs, so no NAT going on or anything like that. However, a couple of the carriers we're trying to use don't like seeing the IP address of the machines initiating the call (in Route and Contact headers, etc.) and that's causing problems including some carriers don't think the call has set up properly (even though it goes through), which leads to missing BYEs. Anyway, seems like topology_hiding() is a great idea anyway, regardless of the fact that I've had a carrier specifically request it. I'm using 1.10. So I've started with the basic Residential scenario made from osipsconfig. I didn't check any of the options (like ENABLE_TCP, USE_ALIASES, etc.) and modified only my IP address and added a couple of aliases for the machines making the calls. I added the following outside of the routing logic to load the dialog module to make topology_hiding() available. loadmodule "dialog.so" Then, under "if(has_totag()) { if (loose_route()) {" I added -- if ($DLG_status==NULL && !match_dialog() ) { xlog(" cannot match request to a dialog \n"); } And outside of the "if(has_totag())" section I added -- if (is_method("INVITE")) { create_dialog(); topology_hiding(); } Without these added sections, things are fine on some carriers and with other carriers I have the problems described above which causes me to want to enable topology hiding. With these added sections, I get 408 timeouts since it appears that the opensips box is responding NOT HERE to the carrier's 200 OKs. Also, possibly unrelated, in either case I'm getting a weird "\304" added to my Route header. Should I just replace the Route header and regex that out? Route: <sip:xx.xx.xx.xx:\304;lr> Thanks! -BJ Quinn ------------------------------ _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 67, Issue 1 ************************************ _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
