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Hello! It's me again. I began a fairly large modification to my CDR proposal some weeks ago, and finally yesterday and this morning got enough accomplished to allow a commit and some peer review. Check the docs out via " svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs " This is a directory; in it you will find: CDRfix2.rfc.doc CDRfix2.rfc.docx CDRfix2.rfc.pdf The docx

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

pose some interesting questions. As a quick and dirty estimate if you assume one of the standard small ec2 instances could cope with 100 simultaneous g711 calls (I don't know if that is the case just guessing) then you'll chew up approx. 2MB/s (you pay for bandwidth both ways). Assuming that you'd then have 1MB/s average to account for quite and busy call times then it would be

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

Greyman-- I'm taking this discussion to the list. Folks, what we are talking about here, is me trying to get a grasp around Greyman's (Aaron's) request for a bare-bones CDR generation that describes just total connect time for channels, stripping out all the details. Who cares about xfer, park, hold, etc.? So in the following is our discussion about what *should* be there, and in

your Asterisk server at all. Try doing a packet trace on the network segment where the calling SIP agent is and see where it's trying to send the ACK to. My guess would be your firewall is incorrectly handling the SIP messages. Generally it's very bad news to use an ALG or firewall to mangle SIP packets as they almost always get it wrong. In your case there is a Record-Route header in the

straight away I can think of a scenario that would cause a problem. If I supply toll free numbers that need to be billed for incoming calls and that can be forwarded out to billable destinations then I want a CDR for both ends of the bridge. In your first blind transfer example what if the initial incoming call to A is billable? I can't see any easy way to get the duration of that call leg.

correct for a transfer. In the traditional Telco World the src (or A Number) field tends to be both the callerid of the customer and an identifier that ties the CDR to the customer for billing purposes. With Asterisk and a lot of other modern day softswitches there's usually a field called accountcode or similar which can be used to tie a CDR to a customer. The src field is then only

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

The Wine development release 1.1.12 is now available. What's new in this release (see below for details): - Some simple 64-bit apps should now run. - Support for subpixel font rendering. - 64-bit code generation in the IDL compiler. - New version of the Gecko engine. - Various bug fixes. The source is available from the following locations:

Hi list, I'd like to raise the issue discussed in the thread below. I've faced exactly the same problem and came to exactly the same way out - shutdown winbindd and use "force unknown acl user". We use "simple" mapping in the environment - all AD accounts have corresponding NIS accounts (the same name) and the mapping is being done by smbd. The problem here is that

On Tue, Sep 03, 2019 at 09:55:49AM -0400, Michael S. Tsirkin wrote: [..] > What's with all of the TODOs? Some of these are really scary, > looks like they need to be figured out before this is merged. Hi Michael, One of the issue I noticed is races w.r.t device removal and super block initialization. I am about to post a set of patches which take care of these races and also get rid of

Thanks Ben, Adam and Petr for the feedback! So currently things that need to be done for the Monitor resource are: 1. Name files uniquely. Adam, your naming suggestion is great. I think we should stick with that, with a minor change: I don't think we should put destination channel name in the file names. In some instances there will be no destination channels (plain IVR: play, record, dtmf),

Mahmut, First of all, I'd like to reiterate what a great patch this has been. I'd also like to voice support for having an option to mix the files on the fly, and name them uniquely. While I was able to smoothly put the files together with soxmix, I see on the fly mixing as hugely beneficial to an automated solution to saving/delivering the messages without intervention. One feature,

RINGING tone. Looking further in the Asterisk logs I see that similar "does not support indication" entries for SIP devices for both indication 3 (RINGING) and 8 (CONGESTION) occur. However for SIP devices Asterisk seems to be able to emulate the ringing and congestion tones properly: Jul 19 16:12:43 VERBOSE[2116] logger.c: -- Executing Congestion("SIP/1000-0820fc88",

The Asterisk Development Team would like to announce the release of Asterisk 13.26.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.26.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

[priv] type=friend dbsecret=dundi/secret context=longdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann <jmann at txhmg.com> wrote: > > > >

Back on the server, I then edited the picture with xv. I used linux, not samba, to call up the image in xv. I then reinserted the image in powerpoint but got the old picture. Ergo, windows had cached the file and linux did not inform samba it was updating the file, so samba let windows continue to cache it. I then did the same excercize, but this time using samba to open up and save the file with

The Asterisk Development Team would like to announce the release of Asterisk 16.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The long term plan is to get all of Dovecot disk I/O asynchronous. The first step to that direction would be to make dbox/mdbox I/O asynchronous. This might also allow mbox/maildir to become partially asynchronous. http://www.dovecot.org/list/dovecot/2009-December/045481.html already started describing how the lib-storage API could be changed to support high-latency storages. Adding support for