similar to: length() of vector after using strptime()

Instead of using a PCI card is it possible to use an outside SIP service for "CO" lines? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030808/33b131e3/attachment.htm

On Wednesday 29 October 2003 18:06, Ricardo wrote: > Hi Mike > > cat /usr/ports/audio/icecast2/distinfo > MD5 (icecast-2.0-alpha-2-20030811.tar.gz) = > fc1dc4173a81daa741cef14cadbf6ce1 > > cat /usr/ports/audio/ices/distinfo > MD5 (ices-2.0-Beta2-20030811.tar.gz) = 5ccf54d6961eba9ac914090dffb874cc > > Should I expect problems with these versions? Is the problem with

I am using spss.get to import an SPSS database "Data.sav", created with SPSS 14 : df1 <- spss.get("C:/temp/Data.sav" , lowernames=TRUE, datevars = c("dateinte")) I am getting this warning. I get the same warning with read.spss. Warning message: C:/temp/Data.sav: Unrecognized record type 7, subtype 16 encountered in system file This is a stupid

Hello I am a newbie to Asterisk. We have set up Asterisk on a PC with Redhat 9.0. We have installed H323 openphone on our PC's. We are wondering what a gatekeeper does. It seems we need one but what I have seen in this group is that a gatekeeper must be installed on another box on the network. As all our PC's on the network use Microsoft OS is there a free gatekeeper software for

Does anybody know why I can NOT hear the MusicOnHold - using SJphone on another PC in our network (normal playback is not a problem) . See the * output and the line configured in extension.conf below (also mp3player does not function) Any suggestions? *Asterisk output:* *CLI> -- Executing WaitMusicOnHold("SIP/jeroen-bf54", "30") in new stack --

Hi folks. I discovered this odd behaviour a few days ago while playing with SAMBA_3 for the first time. The following points apply to samba-3.0.0beta and to cvs-20030807. 1. (tested with tdbsam) If I create a user entry with pdbedit, the domain name field is not set to the domain specified in the configuration file. For me it's getting set to the local machine name. If I logon to a

Can someone in Asterisk'land subscribing to 800 service explain to me how to setup extension.conf to route calls based on the incoming DNIS/ANI. For example I want to route 3 incoming 800# coming across a trunk group to all land in the same queue. So I guess I am asking how to perform DNIS/ANI based call routing? -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Sun, Aug 10, 2003 at 02:31:40PM -0400, jared jennings wrote: > On Fri, Aug 08, 2003 at 09:20:09PM -0400, Chris G (Moguta) wrote: > > >I maintain such a page here: > > > > > >http://thebasement.org/sound/vorbis_info > > > > > >It's probably woefully incomplete. > > > > Well, there are only a few more, AFAIK: > > - MacAmp

hi We are using snome 200 IP phone with *. It works OK. But after a period of time we can't hear any sounds for any icoming or outgoing calls. I've got two of these phones. Same symptoms occur to both of these( not at the same time ) and the problem remains until the phone is completely rebooted. Don't know whether this's * or Snomes' prob. Any help would be appreciated.

Hello, I am looking for hardware for Asterisk. I want to connect analog lines (from 6 to 12 or more) to Asterisk, what will be the best hardware for that? Thanks, Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030819/f4b31f82/attachment.htm

This comes from the Indymedia global audio list. How about setting up a software page on icecast.org with a list of players that work properly? Many players on vorbis.com will only play files on a hard drive, not streaming files, such a list would be very helpful towards promoting Ogg as a useful streaming format. ----- Forwarded message from nathaniel t <vsea75@hotmail.com> ----- The

On Wednesday 29 October 2003 17:29, Ricardo wrote: > Excellent! > > With WinAmp 5 it works but after about 45seconds I get the same messages in > error.log > > [2003-10-29 08:38:33] DBUG format/format_generic_write_buf_to_client > Client had recoverable error -1 This should never happen. It looks like a bug present in some old versions (> 6 months old), are you sure a

On Fri, 8 Aug 2003, Maik Schmitt wrote: [...] > I just tried to use it with our 7960 (sip-version). > > I've set the services_url in SIPDefault.cnf to > "http://xxx.xxx.xxx.xxx/xmlservices/vm/index.php&amp;user=1234&amp;pin=1234" > > It didn't work with ?user=...&pin=.... cause the phone then tried to > get

I should also mention that Accounting rules are not stateful -- each rule only handles traffic in one direction. So for example, if eth0 is your internet interface and you have a web server in your DMZ connected to eth1 then to measure HTTP traffic in both directions requires two rules: DONE eth0 eth1 tcp 80 DONE eth1 eth0 tcp - 80 Associating a counter with a chain allows for aggregation.

Stan, Your patch to oggenc that enables it to read FLAC files no longer patches cleanly on current CVS, but only because oggenc/Makefile.am is different. I put together a naive replacement that looks like it resolves the changes, but then running ./autogen.sh produces the following warning: oggenc/Makefile.am:13: HAVE_LIBFLAC does not appear in AM_CONDITIONAL ...but otherwise completes

Hi Mike cat /usr/ports/audio/icecast2/distinfo MD5 (icecast-2.0-alpha-2-20030811.tar.gz) = fc1dc4173a81daa741cef14cadbf6ce1 cat /usr/ports/audio/ices/distinfo MD5 (ices-2.0-Beta2-20030811.tar.gz) = 5ccf54d6961eba9ac914090dffb874cc Should I expect problems with these versions? Is the problem with Icecast2 or IceS? Ricardo <p>----- Original Message ----- From: "Michael Smith"

All, During development of a backup solution with rsync I experienced some failed backups because of RC 23, partial transfer. These were because the application using the data I was backing up, was still active, and had deleted a file inbetween rsync compiling the file list and then transfering the file. After some feedback from Wayne Davidson and JW Schultz suggesting the solution to this,

We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon

The past week or so I have been experimenting with Asterisk and overall find it to be a nice software suite, although I have encountered some problems, and have found almost no documentation (For example in sip.conf I needed the commands fromuser= and fromdomain= and only figured out this was possible after spending a few hours browsing on the internet and reviewing some person's configuration

On the advisory about the realpath problem it says that it was corrected: RELENG_4_8 src/UPDATING 1.73.2.80.2.3 src/lib/libc/stdlib/realpath.c 1.9.14.1 src/sys/conf/newvers.sh 1.44.2.29.2.2 I ran cvsup and when I look at my src/lib/libc/stdlib/realpath.c I see