[OpenSIPS-Users] Opensips 1.9 getting crashed after successful start
Chandra Prakash
chandraprakash at virtualemployee.com
Sat Feb 1 11:32:36 CET 2014
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
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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" <dlee at digium.com>
Subject: Re: [OpenSIPS-Users] CDRTool - Rating Origination and
Termination differently
To: OpenSIPS users mailling list <users at lists.opensips.org>
Message-ID: <7803ACF4-C5E9-47D7-BE80-110B53095542 at digium.com>
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 <tijmen at ag-projects.com> 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 (tijmen at ag-projects.com)
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 (duane.larson at gmail.com)
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 <duane.larson at gmail.com>
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 at 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
>>> a18cd3be449dd at 108.166.97.17:5060 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 <duane.larson at gmail.com>
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 <tijmen at ag-projects.com>
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 (dlee at digium.com) 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 <ag at ag-projects.com>
wrote:
>>>>
>>>> > I think it would be a good idea.
>>>> >
>>>> > Adrian
>>>> >
>>>> > On 22 Jan 2014, at 16:58, David Lee (digium) <dlee at digium.com> 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
>>>> >> Users at lists.opensips.org
>>>> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>> >
>>>> > _______________________________________________
>>>> > Users mailing list
>>>> > Users at lists.opensips.org
>>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>> _______________________________________________
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>>>
>>> _______________________________________________
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>>>
>>>
>>>
>>>
>>> --
>>> --
>>> *--*--*--*--*--*
>>> Duane
>>> *--*--*--*--*--*
>>> --
>>>
>>>
>>>
>>> --
>>> --
>>> *--*--*--*--*--*
>>> Duane
>>> *--*--*--*--*--*
>>> --
>>>
>>>
>>>
>>> --
>>> --
>>> *--*--*--*--*--*
>>> Duane
>>> *--*--*--*--*--*
>>> --
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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------------------------------
Message: 2
Date: Fri, 31 Jan 2014 18:26:53 -0600 (CST)
From: BJ Quinn <bjquinn at seidal.com>
Subject: [OpenSIPS-Users] topology hiding
To: users at lists.opensips.org
Message-ID: <14008015.29239.1391214413811.JavaMail.zimbra at seidal.com>
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
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