similar to: Most Reliable Proxy Server?

I have a x100p card that has call waiting on the line comming into it and then into *..... is there any way i can use call waiting on that line? Michael

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show

Hi folks, So I''ve got this app built on prototype, with a bunch of extra JS classes to do what we want. Now the problem is, when I make a change to one of those JS files (or an upgrade to prototype itself), how do I ensure that the client browsers refresh their cache with the latest JS? Clearly I want the browse to cache the large JS files, but I also want that cache to expire at

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

I have quite successfully set up the Services button to work on the 7940 running SIP. I have a metric-imperial converter, a foreign exchange rate calculator, a calendar etc available to users. The XML is really fussy though. Simon -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Rich Adamson Sent: Thursday, 1

Hi, Now that I have Asterisk up and running I would like to find a good SIP gateway in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I am looking for peoples recommendations. Apologies if this is the incorrect forum for this type of request. Regards, -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint:

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Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott

Hello everyone. I''m writing web aplications using AJAX (library prototype and scriptaculous). I have situation that I want to change content of one div in my site. I make ajax request to server and getting new content of that div. The problem is that this div should have javascript code (e.g. looks like that <div><script>js code</script>

> Or... > > var myTDs = new Array(); > $A($("main").childNodes).each(function(tr) > { > $A(tr.childNodes).each(function(td) > { > myTDs.push(td); > }); > }); You''ll probably want to make sure the tr elements are TR tags and the same for the td''s, as empty text nodes are inserted randomly by the gecko engine (and maybe others). Greg

Hello, I''m very new to ajax and was looking for advice on scriptaculous at their website and they suggested i join your mailing list, I hope that was the right thing to do, I haven''t installed ROR yet but i have downloaded the needed one-click installer which i will setup sometime soon, until such time i would like to work with scriptaculous in its native form in php files, the

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I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank

I know that the wiki has an extensive list of European VoIP providers out there....but there's so many that it's kind of hard to sort through. So I was wondering if anyone could recommend some reliable SIP/IAX termination providers in Europe? Something like VoicePulse Connect, NuFone, Vitelity, or Junction Networks based out of Europe. I really don't trust a US VoIP company for

On 6/5/2015 1:33 PM, Michael Hennebry wrote: > It's a desktop in an old house. > The outlets have ground-fault protection, > but the third prong is ungrounded. > not sure how GFI would function at all without a valid ground, unless the GFI is wired to neutral, which is dangerous on its own. you might get a 3-prong-to-2-prong adapter and plug the PC into that, leaving the ground

On Fri, 5 Jun 2015, g wrote: > On 06/05/2015 03:43 PM, John R Pierce wrote: >> On 6/5/2015 1:33 PM, Michael Hennebry wrote: >>> It's a desktop in an old house. >>> The outlets have ground-fault protection, >>> but the third prong is ungrounded. >> >> not sure how GFI would function at all without a valid ground, unless >> the GFI is wired

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP

Unfortunately I seem to have another problem! I am using sipgate for the incoming line - and it appears that you cannot get DTMF to work in that configuration. Unless anyone knows anything different of course!! > > I just want one of my incoming numbers to go to an IVR service that > > will allow me to select what I want. > > > > For example > > > >