similar to: Anyone know a good ITSP in Canada that suppo rts *?

Hi, I'm looking for a new Internet Telephony Service Provider for my company in Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in Otawa, Toronto, NY & San Jose. Anyone out ther who can help me with a recommendation? Vonage seemed clueless when I called them. Broadvoice is good but no Canadian DIDs... Thanks, Hugh -------------- next part -------------- An

If you would like to buy cheap WOW Gold (http://www.storeingame.com/) and WOW Power Leveling (http://www.storeingame.com/level/World-of-Warcraft-EU/power-leveling/), we can offer you anytime you want. Knowing all the secrets of a zone, I get to manage properly every minute of any of my gameplay sessions. Since the Aion Power Leveling (http://www.storeingame.com/level/Aion-EU/) guide I have been

I don't know exactly the appropriate list to ask this question in but the web page gave me the impression this was. I don't particularly want to subscribe to the list as I can't handle the volume of messages which I'm not concerned with. I would just like to be able to ask a question concerning the email subject and receive ONLY replys to that message. I'm not too familiar with

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

Is plot(ts.object) equivalent to the S-Plus ts.plot ? -- ------------------------------------------------------------------------- Konstantinos E. Vorloou | Tel: +44 (0)191 374 1821 Department of Economics & Finance | Fax: +44 (0)191 374 7289 University of Durham, | email: K.E.Vorloou at durham.ac.uk 23/26 Old Elvet, | or : vorlow at rocketmail.com

wahh :( :( so how can you get some demo files?

NICE !!!!!!!!!!!!!!!!!!!! im here waiting for the progress admin :) tnx i appreciated that you trying to help me :( :) :) TNX :) :) :)

That's a good question. never heard it before..!!

There's nothing really noteworthy in that log, I'm afraid. The best way forward is for a wine developer to get a copy of this software so s/he can see the error for themselves, I'm afraid. And even then it's not clear they'll be able to fix it. You're not quite the first person to ask about this: http://community.sageaccpac.com/forums/p/189/571.aspx And about four years

Tok tok!!! Still waiting "_"

Fr??d??ric Delanoy wrote: > On Tue, Aug 9, 2011 at 05:28, joven15 <wineforum-user at winehq.org> wrote: > > > Clang clang !! Still waiting :( > > > > We know you're still waiting. You've made that clear many times already. > Please don't do that all the time: it just annoys people and won't > speed up the resolution of your issue. Agreed.

Hi Not sure if this is going to the right place… I am attempting to use a S-Plus workspace that has been saved via the dump facility. These include a number of commands not recognised by R, namely rts commands. Is there a way round this by changing these in some way for R’s ts commands? Any help much appreciated. Adam [[alternate HTML version deleted]]

Hello, I'm trying to transform reaction times which are not normally distributed to an ex gaussian or an inverse gaussian distribution, but I don't really know how to use the exGAUS() function. Can someone show me a script in which data has been transformed? Thanks in advance k [[alternative HTML version deleted]]

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

We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. ?To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b66661d72f1328c918@%EXTERNIP% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b66661d72f1328c918@%INTERNIP%. ?I can

Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak michiel at vanbaak.eu http://michiel.vanbaak.eu GnuPG key:

All, I am trying to find a SIP ITSP that honors dial around compensation. We are adding a Flex ANI code to our outgoing SIP invites by appending an isup-oli tag to our From: address, like this: INVITE sip:18889996563@carriers.icall.net SIP/2.0 Via: SIP/2.0/UDP xxx.y.34.201:5060;branch=z9hG4bK7f314484;rport From: "Dougs Payphone"

We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. ?To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b66661d72f1328c918@%EXTERNIP% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b66661d72f1328c918@%INTERNIP%. I can

Anybody know of a VoIP (preferably IAX) carrier here in the Atlanta area? Using ones in Chicago and Denver is hit and miss these days. Traceroutes seem to change often and the latency is inconsistent. I'm wondering if someone more local would improve things. -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC paul@dugas.cc phone: 404-932-1355 522 Black Canyon

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I