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...er!" (very annoying!) To configure the new phones you have to press and HOLD the up arrow whilst plugging in the power/ethernet, you should then get the option to select SIP. will let you know how I get on with the 5220 with asterisk as and when I get my phone :-) Sam Colin Anderson <ColinA@landmarkmasterbuilder.com> wrote on 01/09/2004 16:36:24: > > > > I will get a packet sniffer on one in a minute.... > > Don't bother, won't work. I already tried. Spoke to some Mitel mucky-mucks > too, and they said nope. You have to get Mitel's SIP-s...

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

...nction.to.evaluate.string(x) } As a result, I would have n data.j objects I've been looking for a function in R to evaluate a string and I haven't found anyone. Is there any function to do that? Thank you very much -- ============================================================== Alvaro Colina |-|o||o||o||o||o||o||o||o||o|- Area de Quimica Analitica | Pza. Misael Banuelos s/n Facultad de Ciencias | 09001. Burgos. Spain Universidad de Burgos | Phone: 34-947-258817 e-mail: acosa at ubu.es | FAX: 34-947-258831 ============================================================== -.-.-...

...as 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: asterisk-users@lists.digium.com Subject: Re: SNOM 190 '486/Busy here' after upgrade to re 3.60s Hi Colin, There are a few reasons why a phone would deny a call with reason=busy: 1. If redirection is somehow on without a redirect target set. An incoming call...

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

...ontal [1] TRUE $width [1] 5 $height [1] 5 $family [1] "Helvetica" $pointsize [1] 12$bg [1] "white" $fg [1] "black" $onefile [1] TRUE $print.it [1] FALSE $append [1] FALSE Thanks in advance, ============================================================== Alvaro Colina |-|o||o||o||o||o||o||o||o||o|- Area de Quimica Analitica | Pza. Misael Banuelos s/n Facultad de Ciencias | 09001. Burgos. Spain Universidad de Burgos | Phone: 34-947-258817 e-mail: acosa at ubu.es | FAX: 34-947-258831 ============================================================== -.-.-...

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

...hing.... Thanks, -- -- -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: caloi@usadatanet.com <mailto:caloi@usadatanet.com> -- -- -- _____ From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Friday, February 17, 2006 10:42 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] A unique 'click to call' project - Could usesome advice You create a context in your dialplan that accepts the DID to ca...

...9; SIP service can work with Asterisk. We tried setting it up like how you would for iconnecthere However, we even failed to register in the first place! (Of course password and username are correct). Anyone else on the list successfully used Primus' SIP with Asterisk? David >>> ColinA@landmarkmasterbuilder.com 1/20/2004 12:25:50 PM >>> Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html...

...9; SIP service can work with Asterisk. We tried setting it up like how you would for iconnecthere However, we even failed to register in the first place! (Of course password and username are correct). Anyone else on the list successfully used Primus' SIP with Asterisk? David >>> ColinA@landmarkmasterbuilder.com 1/20/2004 12:25:50 PM >>> Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html...

> 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

...great. Thanks, -- -- -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: <mailto:caloi@usadatanet.com> caloi@usadatanet.com -- -- -- _____ From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Friday, February 17, 2006 12:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] A unique 'click to call' project - Could usesome advice Same as before but instead of SIP as the origination channel you...

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

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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