similar to: Monitor application

Hi all, I'm using asterisk in a production environment now and this afternoon I got reports complaining that it was not working. Looking at the asterisk console output, I saw it contains lots of error messages as printed below. Unfortunately it is not obvious from the logs as to what started all this. Just before the error messages start, everything seems to be working fine with no problems.

Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

Hi, in the Snom FAQ I found the following information: After staring up, the phone tries the URL given in the "Setting URL" of the phone. ... BTW this setting can also be set via DHCP. .... option tftp-server-name "http://192.168.0.9/snom200{mac}.htm" The documents used: FAQ-04-06-14-sf.pdf "Setting up DHCP for snom phones" FAQ-04-03-24-sf.pdf "How can I

Some of you have got Wifi600 wireless SIP phone working with Asterisk. Specially John Todd ( nice review ). My phones register ok. They can also receive calls from other phones. But for some reason I can't make them call out ( anybody, ie. SIP or PSTN ). This seems to be due to the phone not understanding what it should do when it receives 'Proxy Authentication Required'. In my case

Has anyone done rollbacks with puppet? The story goes that last night I pushed a change out, which intentionally changed the way certain services where managed. The change involved pushing out a few files, cronjobs, etc., which was okay until it was discovered there was a bug with the service. My configurations are kept in subversion and rolling back to a previous configuration would have

Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying

Just change the install prefix directory in asterisk makefile to something like /usr/local/asterisk, and then you will see all the files needed for asterisk to run. > -----Original Message----- > From: Steven Critchfield [mailto:critch at basesys.com] > Sent: Saturday, 1 March 2003 22:03 > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] How could I install

These are date and times in the format YYYYMMDDhhmmss. I would like to take this column and make a chron object form them. I have tried a couple of the split family of functions but they need character input here is the data: date.time <- c(19851001001500, 19851001003000, 19851001004500, 19851001010000, 19851001011500, 19851001013000, 19851001014500, 19851001020000, 19851001021500,

Hi all, I found that the figure returned by MSet::get_matches_estimated() varies depending on how results are to be sorted. For instance, in my index, value 4 contains date and time in the format "yyyymmddhhmmss". For the same query, the number of results will be estimated to 20000+ when results are first sorted by date and time with set_sort_by_value_then_relevance(4) and to only 100

Mahmut, excellent summary :-). I look forward to your next update. One little thing, In the manager events that show start/stop monitoring, can you please include a field that indicates the filename(s) to which the monitoring was written? Thanks, Ben -----Original Message----- From: Fettahlioglu, Mahmut [mailto:Mahmut.Fettahlioglu@oa.com.au] Sent: Tuesday, April 15, 2003 5:17 AM To:

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

This is automatically generated email about markup problems in a man page for which you appear to be responsible. If you are not the right person or list, please tell me so I can correct my database. See http://catb.org/~esr/doclifter/bugs.html for details on how and why these patches were generated. Feel free to email me with any questions. Note: These patches do not change the modification

[apologies for no line wrap; config lines at bottom] I have mentioned on several threads here that the Monitor application doesn't do exactly what one would expect: the originating and answering legs of a call are unsynchronized by the duration of the interval that it takes for the answering leg to pick up the phone. This can be very distracting in a final mixed version of the file. Brian

Here's the latest batch of words to get shipped out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket. Please send to paypal address "jtodd@loligo.com". I did not include all

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,

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),

Thanks to Digium for the fine work they are doing on queue app! Does anyone know the progress on the monitor app recording to one file or synchronizing the two files' start and end times? Thanks. Jim Friedeck

This came up when trying to compile the Python bindings against llvm 1.7. File include/llvm/Transforms/Scalar.h declares llvm::createPREPass(), which does not seem to be implemented anywhere. Should I report this in llvmbugs? -- Pertti

Hello - Here's my first impression review of the first SIP 802.11 phone. I got my hands on the "first" one sold, so that perhaps makes me the first person to have a real 802.11 SIP phone commercially in the US interworking with Asterisk. Whee! Can someone point me to other commercially shipping phones to prove me wrong?