similar to: Redirecting to another service/server

Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and find that it's extremely slow for configuring. For instance, it takes several minutes to boot up, apply any changes via the web interface takes at least a minute, etc. Is this normal behaviour? Is there anything that can be done about it? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591

Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over ethernet and doesn't require any authentication, what sort of a trunk would need to be created? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality

Hi guys. I'm trying to disable all debug output, but am not having any success: nick@asterisk-dev1:~> sudo asterisk -r Asterisk 1.2.8, Copyright (C) 1999 - 2006 Digium, Inc. and others. <..snip...> certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.8 currently running on

Hi guys. I just bought and configured a Snom 360 and have noticed that the LCD is constantly flickering at a rate of ~10-15Hz (that's a guess). Either way, it's very distracting. Has anyone else encountered this before? Any solutions? Cheers, -- Nick E: nick.hoffman@voxpak.com P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make

Hey Everyone, We are in the process of reviewing headsets for use with our GXP-2000s. I'm looking for some feedback as to which headsets people are using, the pros and cons of those headsets, and if they would recommend them to someone else. Any help would be appreciated... - Jason -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys. I've found a fair bit of information with regard to how to setup Asterisk to send and receive calls through SER (SIP Express Router), however I can't figure out what information is up-to-date. Many of the examples suggest using insecure=very for the SER entry in sip.conf, but I believe that option is for pre-1.2. If you would care to share which settings work best for adding

Hello all, I want to sniff all these info to test a sip ip phone talking to a asterisk server. I have used tcpdump, but It just shows the UDP, length: 602 Anyway to see the sip uri. Host info? Regards, Dinesh. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

Does anyone know if a TDM2400 will fit into a Dell 2850? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com -------------- next part -------------- An HTML attachment was scrubbed...

Looks like I am going to be doing my first serious commercial install of FreePBX. I DO mean serious. They are willing to put up with a few glitches initially which is why I have decided they will be a good first client. I have turned down several over the past couple years because I just did not feel comfortable with the state of software/hardware. It seems to work much better now. I was

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who dialled. I need to

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

I was encouraged to post this notice on both asterisk-users and asterisk-dev; sorry if this is overkill, but it **is** applicable to both communities. Since the report is fairly large, has a pretty graph, and the whole bit, it was thought that posting on it a website, and letting you browse it would be better than sending hundreds a 50K message. http://www.asterisk.org/SpeedvsSizeExperiment

Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR

On 3 Aug 2015, at 7:16, Steffen Kaiser wrote: > On Mon, 3 Aug 2015, Paul Hoffman wrote: >> On 2 Aug 2015, at 23:27, Steffen Kaiser wrote: >>> On Sun, 2 Aug 2015, Paul Hoffman wrote: >>> >>>> Greetings. I have a bunch of mbox files that I want to import to an >>>> existing user, each into a new mailbox. That is, I want to import >>>>

On 2 Aug 2015, at 23:27, Steffen Kaiser wrote: > On Sun, 2 Aug 2015, Paul Hoffman wrote: > >> Greetings. I have a bunch of mbox files that I want to import to an >> existing user, each into a new mailbox. That is, I want to import the >> file project1.mbox to user Foo, into a newly-created mailbox called >> "Project 1". Is there a dovecot tool to do this

Dear readers, S-Plus have a rts function (Regular Time Series), which is used like so: fveks<-read.csv('http://louise.hoffman.googlepages.com/veks.csv',header=TRUE,sep=',') attach(fveks) acf(ts.intersect(rts(HC.f),rts(Ta.f),rts(GR.f),rts(W.f))) Warning the csv file is 750kB. Can the same be done in R? Lots of love =) Louise

On 9 Aug 2015, at 8:46, Benny Pedersen wrote: > Paul Hoffman skrev den 2015-08-09 17:40: > >> Initializing mail storage from mail_location parameter failed: mbox: >> mbox root directory can't be a file >> >> Which bit am I missing here? > > sudo doveadm import -u Foo mbox:/home/phoffman/project1/ "Project 1" > all > > untested When I

Dear help list, I have a timestamp in as a chron object: > (x <- chron(dates = c("12/02/11", "22/11/11"), + times = c("07:30:00", "04:00:00"), + format = c(dates = "d/m/y", times = "h:m:s"))) [1] (12/02/11 07:30:00) (22/11/11 04:00:00) Now I want to shift the timestamp by 06:30 (hh:mm) backwards, to get