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...na(StoreData$P2)) FALSE TRUE 2 940 0 3 0 323 4 288 0 5 306 0 > table(StoreData$CompanyID, + !is.na(StoreData$P5)) FALSE TRUE 2 940 0 3 0 323 4 288 0 5 306 0 Here's the loop that I wrote, which does not work: angus <- c(2,5) for(i in angus) { cut <- paste("StoreData$P",i, sep="") table(StoreData$CompanyID, !is.na(cut)) } When I run the above, I get the following error message: Error in table(StoreData$CompanyID, !is.na(cut)) : all arguments must have the same length >...

...00/200 192.168.0.3 D 255.255.255.255 5060 Unmonitored 200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100 IP phone. relevant bit of sip.conf: [200] username=200 type=friend secret=1234 port=5060 nat=never dtmfmode=rfc2833 context=default callerid="Angus Comber" <200> host=dynamic disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 [202] username=202 type=friend secret=1234 port=5060 nat=never dtmfmode=rfc2833 context=default callerid="Sam Comber" <202> host=dynamic disallow=all allow=ulaw allow=alaw allow=g723.1...

...tem with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and deleted immediately? Angus Comber angus@NOSPAMiteloffice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050720/c9499a7a/attachment.htm

On Tue, Jun 6, 2023 at 2:19?PM Angus Chen <angus.chen at jaguarmicro.com> wrote: > > Hi,Jason. > > > -----Original Message----- > > From: Jason Wang <jasowang at redhat.com> > > Sent: Monday, June 5, 2023 2:54 PM > > To: Angus Chen <angus.chen at jaguarmicro.com> > > Cc: mst at...

...300 get: WARNING[22077]: channel.c1970 ast_request: No channel type registered for 'IAX' NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of type 'IAX' I have restarted Asterisk after config change. What have I not done. I am just testing the iaxComm program. Angus

...I am trying to display / print certain columns in my data frame that share certain condition (for example, part of the column name). I am using for loop, as follow: # below is the sample data structure all.data <- data.frame( NUM_A = 1:5, NAME_A = c("Andy", "Andrew", "Angus", "Alex", "Argo"), NUM_B = 1:5, NAME_B = c(NA, "Barn", "Bolton", "Bravo", NA), NUM_C = 1:5, NAME_C = c("Candy", NA, "Cecil", "Crayon", "Corey"),...

On Tue, Apr 11, 2023 at 06:36:39AM +0000, Angus Chen wrote: > Hi. > > > -----Original Message----- > > From: Jason Wang <jasowang at redhat.com> > > Sent: Tuesday, April 11, 2023 1:24 PM > > To: Angus Chen <angus.chen at jaguarmicro.com> > > Cc: mst at redhat.com; virtualization at lists.linux-fou...

...pgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add anything to extensions.conf? do something else. I also tested with a Snom 190 and that cannot pickup using *8 either! Angus

...ll think that was the problem) and stopped operating as PBX at one stage. Anyone any experience of this sort of thing? any ideas how to fix - ideally I don't want to have to fit a fan. Is SUSE not the best distro to use for this sort of thing? Should it be something to take up with VIA? Angus

...ith a 600MHz Eden Fanless CPU. Is this likely to be enough power for a 8 extension system with 6 external pstn lines? How important is cpu? Is there some measure, eg xMHz CPU per extension or something benchmark? I have installed 512MB memory - again any benchmark for asterisk memory usage? Angus

...I really hope this helps someone else out there because it caused far too much pain for me to want to have gone through it just as a personal exercise. Cheers! P.S. For those of you viewing this as text, the blog entry is at: http://falkayn.blogspot.com/2006/04/playing-with-ruby-on-rails.html -- Angus McDonald -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060409/1515ca1d/attachment.html

...ror: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 I have set a password for root on mysql - could that be the problem? Should I remove the password? What is easiest way to do that? Angus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050721/b659405f/attachment.htm

...voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more expensive phone - eg a Snom to see what they were like for echo. Is there something I can do with the Asterisk? codec to use? Anything? Angus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050805/c379bde2/attachment.htm

...+34<rest of number> I am in UK which is +44 so cannot work out why seeing +34. In my zapata.conf I have: loadzone = uk defaultzone = uk I can't find any country specific stuff in any other conf files. Any ideas how I can correctly set so that calls from my asterisk do not have +34? Angus

...n 1 -- Saved useragent "X-Lite release 1103m" for peer 202 -- Extension '787367' in context 'frompstnisdn' from '07768385144' does not exist. Rejecting call on channel 0/1, span 1 Do I need to enable something to be able to use the s in extensions.conf? Angus

Hello I am using a Snom 190 and the quality seems OK. Trouble is the volume is quite low and full volume on the Snom is still too low. Is there something I can do on the asterisk to increase the volume? Angus

best described here: https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=64689 Confirmed also present with the rpm build at http://rsync.samba.org/ftp/rsync/binaries/redhat/rsync-2.4.6-1.i386.rpm Please cc: me on replies (I'm not on the list, yet - my procmailrc's in a major state of flux as I'm switching machines) and/or add comments to the above bugzilla entry James -- James

...at in his section on the TDM400 on page 127 he says "In the UK, you may need an adapter that provides a ring capacitor, or the phone may not ring." Can anyone confirm this. Also what is one of those and where would I find a good supplier? I am in the trade so wholesale would be OK. Angus Comber -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050608/80171340/attachment.htm

Hello I setup Asterisk@Home with purely VoIP and it worked fine. I then added an X100P card so I could call out / take inbound calls via PSTN and that went fine. But I have just added a TDM400P card (specifically a TDM30B) and now problems. Here is some of the output. Any ideas on what I should be looking at next? When I run genzaptelconf -s -d I get lots of erors on screen - bit I can