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hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

sorry for the cross post, but this is germane to the developpers as well as the larger user community. Re: [SIP 0000104]: [patch] Cisco-like NAT trick for outbound SIP connections On Sat, Jan 03, 2004 at 08:07:29PM -0600, bugs@digium.com wrote: > > A BUGNOTE has been added to this bug. tabarnac! it's been months now! the only thing that i can think at this point is that mark

...a post-1.0 tag in the bugtracker since there seems to be lots of interest in it's existance, but it needs a little work. Please do not make assumptions as you have below, because I am the author of this patch, and I do *NOT* feel as though i'm finished, nor did i say so anywhere in the bugnotes. Thanks. twisted box100 wrote: > Can anyone tell me how I can implement the features added in the > following link for call transfer? The authors seem to feel they are > finished but it doesn't appear to have been integrated into what > everyone can download. It is referred to as...

Hello - As I've mentioned before, we have a large number of patches starting to build up in the bugtracker interface waiting for addition to CVS. Many of these patches are ready to go and have been fully tested but have not been added due to time schedules. However, almost all of the bugnotes that have the term [patch] in the title have code in them that has been submitted but does not have any verifiable testing done on it other than what the author has done. WE NEED TESTERS TO CONFIRM PATCH VALIDITY. The folks at Digium (mark, specifically) don't have time to do exhaustive...

I've submitted http://bugs.digium.com/bug_view_page.php?bug_id=0000434 requesting that Digium put up a page with links with external Asterisk related resources. If you have a web site with Asterisk related information, patches, samples, documentation, etc, please add a bugnote to the above URL. There is a lot of good information out there, but time and time again I hear complaints that

In response to the SIP and NAT discussion, I have updated the ticket on the subject that seemed to be getting the most attention: #104. There are enough clueful people here that perhaps someone can come up with a patch that handles NAT in the elegant way that I describe in the bugnotes, as I am but a mere integrator who has limited C skills. In the absence of such a patch being offered, we await William Waites' patch and disclaimer which will at least be more sufficient than the current externip= method. Those with an interest in the discussion of how Asterisk should han...

I've been looking at the weather vocabulary in asterisk-sounds in CVS. I've run into a few hitches with words I can't seem to find. So far, I'm looking for 'point' (for constructing floating point numbers) and 'around' as in "high around 70" (don't I wish). Any chance of getting these? While I'm on the subject, I'd be very interested in a

Can anyone tell me how I can implement the features added in the following link for call transfer? The authors seem to feel they are finished but it doesn't appear to have been integrated into what everyone can download. It is referred to as a patch but I don't understand how it could be applied. Here is the link: http://bugs.digium.com/bug_view_page.php?bug_id=0002010 I guess I just

If you run CVS HEAD, and are using ODBC storage for voicemail, please apply and test the patch from bug 3024 in Mantis. This patch should _not_ change the behavior of your system at all, and that's what needs to be tested. If you can try it, please report the results in the bugnotes. Thanks!

Good day all What is the file sip_notify.conf for Thanks Altus

Hello, Just confirmed this on my end, because of the massive changes that have been made to callerID handling in asterisk 1.0.5 many of the features of the astGUIclient suite will not work on this new version. The latest stable version recommended is Asterisk 1.0.3. We will work on trying to find ways around the new callerID rules that the asterisk developers have put in place and hope to have

All, As a bug marshal, I have noticed quite a few bugs that seem to get overlooked/ignored due to the fact that they do not have the appropriate information in the bug information fields, or bugnotes. I am, therefore, Reposting an old post from July 26th, 2003, to help clarify the use of the Bugtracker. Sorry if you have read it before; this is just an attempt to re-familiarize everyone with the information that should be included in the posts, which should help speed things along. Thank...

A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the

I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and line 2 on 5061. The SPA2000 is on IP address 192.168.17.6, and the asterisk box is on 102.168.17.2. Both SPA2000 ports(5060 and 5061) share the same IP address. Every minute I repeatedly get the following output: SIP Debugging Enabled 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.17.6 SIP/2.0 Via: