similar to: [BOOK] VoIP Telephony with Asterisk

Does anyone know if there are any reseller for the book "VoIP Telephony with Asterisk" in Hong Kong/Asia region? I'm interested in purchasing the book but the shipping charge to Hong Kong is expensive. Thanks. Joseph -------------- next part -------------- An HTML attachment was scrubbed... URL:

Just received my copy of this book today. Based on a cursory examination, "VoIP Telephony with Asterisk" (VTwA) by Paul Mahler looks like a very decent book for anyone starting out with Asterisk. More knowledgeable readers may also benefit from perusing its 290 pages as well. Unlike "Asterisk for Small Office Setup" (which I slammed in a review a couple of weeks ago) VTwA has

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

Hi List, I know that the zaptel modules have echo cancellation, but is this possible to do this on VoIP <-> VoIP traffic as well? I'm toying with a SIP gateway which has apparently a terrible call quality and would like to know if there is any way asterisk can help with this. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP

Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel.

Introducing the simple Twitter Directory for telephgony, VoIP and asterisk users, developers, providezrs, manufacturers and hobbyists If you're into Twitter and you'd like to take literally 10 seconds to go fill out this form with two fields, your Twitter name will be in the public directory of people who are interested in these technologies. http://tr.im/voipform You can look at the

Hi List, I've got * randomly hanging up on inbound or outbound calls on zap channels. I use a Digitnetworks X100P clone card. Any idea of what might be happening? Cheers, Jean-Michel.

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi List, After some research, it seems the only reasonable thing to do in order to get SIP phones behind NAT working reasonably well without fiddling with the DSL router is to have some kind of far end nat traversal mechanism. Is there any way to do this with open source tools? I've seen somewhere that far end nat traversal can be achieved with SER + nathelper does the job... has anybody

Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI> Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas?

Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Dear R-Helpers, Apologies in advance as this is partly (widely ?) OT. Not sure where to ask, R is my favorite computer tool (no kidding) and there are plenty of knowledgable and helpful people on the list. Background: There are discussions among experts and regulatory authorities (cf guideline http://www.emea.europa.eu/pdfs/human/ewp/056598en.pdf) that, in for example Amyotrophic Lateral

Hi List, Do you know if there are any plans to improve i18n for Asterisk? The current i18n way of doing it with asterisk is very limited and most of the time does not work. For example, take voicemail: "message" "received" "at" "seven" "30" "am" might sound good in English. But: "message" "recu" "a"

Dear R experts How can you perform a regression analysis in R when the dependent variable is countiuous but bounded, say between 0 and 100? I would be grateful for pointers to R-functions but also for hints to relavant litterature since I have never worked with this problem before. Thanks in advance. Kim Mouridsen. [[alternative HTML version deleted]]

Hi List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi->exec ("DIAL $dialstring"); my $answeredtime = $agi->get_variable ("ANSWEREDTIME"); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? I'm using asterisk 1.2.7.1 and the

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want

We just updated our configuratiosn to have multiple NIS servers, when we initiated a test of client failover, we were disapointed. It seemed that the only way to get a filaover was to /etc/init.d/ypbind restart. It behaves as indicated in http://bugs.opensolaris.org/bugdatabase/view_bug.do?bug_id=5084845 using ypbind-1.17.2-13 on Centos 4.5 / Linux xxxxxxxxxxxx 2.6.9-55.0.12.ELsmp #1 SMP Fri Nov

On Sun, Mar 29, 2009 at 3:33 PM, Benoit Boissinot <bboissin+llvm at gmail.com> wrote: > While it is not described in the litterature, I don't think you need > to introduce a new > function: >      x0 = ... >  x1, x2 = \sigma (x0) >         | >    +----+------+ >    |           | >    v           v >  ... = x1    ... = x2 > > Can be transformed to: >

How can I make a reference i this case? I want to make a reference to 'Artikel XXX' For example In The Artikel 'XXXX' there is two tables. .. Litterature Artikel XXX -- View this message in context: http://r.789695.n4.nabble.com/Make-a-reference-tp4634611.html Sent from the R help mailing list archive at Nabble.com.

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000