search for: colinas

I know this is not strictly an asterisk issue but it is related I guess. Just to let you know that after many calls to Mitel the consensus is that they will be releasing a new version of the 5220 that is dual boot (minet and SIP) next week or the week after. This firmware will only appear on NEW phones manufactured after the release date (no one could confirm but the 23rd of sept was

>I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN?

If I have several character vectors, for example: a_c('data.') b_c('_myfunction(') c_c('vect.') d_c(')') and, j_1 I can build a vector using paste: x_paste(a,j,b,c,j,d,sep="") x="data.1_myfunction(vect.1)" I have n numeric vectors (vect.1...vect.n) Then, if I could evaluate the string x, I would calculate the result of my function in

AHA! # 1 is the case! Seems the user was fooling around with the phone after the firmware upgrade. Shame that that setting couldn't be locked out. Thanks to Mr Tahir and Mr Stredicke for their spot on responses. -----Original Message----- From: Usman Tahir [mailto:Usman.Tahir@snom.de] Sent: Friday, September 23, 2005 12:34 AM To: ColinA@landmarkmasterbuilder.com Cc:

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

I'm using R version 1.1.0 with Redhat 6.2 in a Dual-Pentium When I type postscript() I have the next error: -- Error in old$command == "default" : comparison (1) is possible only for vector types I have updated R next week using the rpm package http://www.stat.math.ethz.ch/CRAN/bin/linux/redhat/6.x/i386/R-base-1.0 .0-2.i386.rpm I know that postscript() has new features, but I

Hello, I would like to remove some files which have the extension .test for example (data1.test, data2.test ....). Is there another solution to remove them instead of doing it one by one ? Thanks for your help, St?phanie Langevin -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send

I'm using a Snom 320 in a CAP position and the receptionist wants to do blind transfers. OK, no problem so far. Now she has asked me how to UN-transfer a call, as in, she transfers a call and wants to hook the call back before it connects (she wanted to tell the caller additional information for example) I don't think that this is possible as once my dialplan starts using Dial()

the user you are connecting as should have full rights to /var/run/asterisk: http://www.voip-info.org/wiki-Asterisk+non-root hth -----Original Message----- From: Erick Perez [mailto:eaperezh@gmail.com] Sent: Monday, April 03, 2006 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk.ctl

Colin, Thanks for your assistance. Reading over your advice I seem to still be a bit confused. My agents are not on the Asterisk server; it appears in your advice that my the call will travel this path: WWW interface --> agent enters their DID, platform to use, and termination DID --> AST calls agent --> Agent calls termination DID If my agents are not on the Asterisk server

I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David >>> asterisk-users@eol.ca 1/21/2004 6:39:34 AM >>> I'm not sure Primus

If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP.

> They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and > 'worst' readings of 99.975586% and 99.963379% respectively. Might want to give PCI latency setting a try, it helped for me. My ZTTEST would drop occasionally to 99.95% until I set: setpci -v -s 01:01.0 latency_timer=ff

Disclaimer: Not a troll I'm curious as to this obsession with uptime is. All of the posts of this type are along the lines of "After X days, Y thing does not work but if I reload or reboot, it's OK" - so why not cron a reboot? Is it considered bad form or something like that? I reboot every night whether it is needed or not, not afraid to admit it, and everything works fine for

In the example I posted previous, there is an obvious gaping security hole, it would be trivial for someone to read the querystring and exploit it to make free phone calls, spoof caller ID (if you allow the CallerID to be set with a QueryString value), etc. You want to make damn sure that the URL is not publicly accessible or somehow obsfucate the querystring, or use POST. In my case, I

I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

http://www.pcmag.com/article2/0,1759,1812887,00.asp Specifically, his assertion that ISP's would sniff traffic and block, say, the SIP port. You could play wack-a-mole with port numbers, no? Also a community based, Freenet style of encryption implementation for "free" VoIP traffic would address this issue. I raise this to the list because I'm sure there's a grain of

...ANCISCO HELSON DE LIMA NERES, PAULO RAFAEL PEREIRA SOARES, JO?O CARLOS MOREIRA DE CARVALHO, DAMI?O JOVENAL DOS SANTOS, MARIA GORETTI LIMA FREIRE, JANIMERY BARBOSA DE ABREU MELO. SHYSLAINE ARA?JO BEZERRA, ARIANE SOARES SILVA, LUCAS MOREIRA DIAS, GILSON POLICARPO DE S?, REBECA DE FREITAS BARROS. Nova Colinas. Jacu?pe, ANNA JESSYCA ANDRADE LACERDA, LUCAS DE PAULA OLIVEIRA, GEORGE FACUNDO RICARDO, RAFAEL WESLEY MENESES MAIA, JO?O CARLOS MOREIRA DE CARVALHO, DEBORAH LIMA BRAGA, MARIA MARIANA RIBEIRO VIEIRA, JO?O VICTOR RODRIGUES DE C. MEDEIROS. TATIANA RODRIGUES DE CASTRO MELO, CAIO URBANO CAMURCA, MAIAR...

I've got a lot of problems saving plots using postscript(file="xx.ps") The quality is not too good when I print it. Is there any way to save good quality plots in a file? Is there, perhaps, another function? I have tied save.plot(), postscript(), ... Thank you very much -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read