similar to: OT: Accountless, free, skinnable, browser based SIP client wanted

I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware

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I want to take series of user entered (via phone keypad) options/numeric entry fields and use these with an AGI script. I have looked through voip-info and I can't find any Asterisk functions specifically for this. Any guidance please? Michael

I'm looking at the <http://support.a-link.com/phonemate/IPU1.htm> phone because it works with Skype (from Linux), but can do SIP, too. Not necessarily asterisk related, but possibly. My networking situation might require IAX if I'm running Linux and want to use SIP, I'm not certain (Skype works fine). Putting that unknown aside for the moment, how does this phone work under

Hi, I've set up a Gizmo Project account for access on my Nokia E61 because they work through NAT. Trouble is If I include my gizmo account in an asterisk hunt group and I'm not connected (phone is off / outside wireless coverage) the gizmo project always answers. Either the call goes to voice mail or if I turn voicemail off the call gets answered by a recording saying I'm not

Got an email this morning with the subject "Welcome to Gizmo Project". I didn't sign up with those yokels. Anyone else got spammed by them?

Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance,

Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file when the user picks the phone. I use Gizmo service for SIP and I'm able to call through it. However, when asterisk dials a number, Gizmo first answers then tries bridging 2 channels. Right after answer Asterisk starts playing the reminder. It obviously results in

Some !@$#@@# in the Czech Republic used one of our SIP accounts to place four thousand calls to what appears to be a toll number in Zimbabwe last night. Filter 82.150.165.5. A more overriding problem for me is how do we know what *destinations* to filter so this idea of war dialing a toll number is something we can cutoff before it gets to our upstream provider? Is there some collected

Hello. I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly

Oops, I just erased all my data using this gizmo that I thought would replace -9 with NA. A) Can I get my tcn5 back? B) How do I do it right next time, I learned my lesson, I'll never do it again, I promise! Anders Corr > for(i in 1:dim(tcn5)[2]){ ##for the number of columns + for(n in 1:dim(tcn5)[1]){ ##for the number of rows + tcn5[is.na(tcn5[n,i]) | tcn5[n,i]

Im New to Linux and Wine. I use Ubuntu 8.10, just downloaded wine. all my drivers are working :). Now, I have the CD's for Warcraft 3, Its stated Installing Fine but half way installing then worldedit.exe and it crashers giving me [img=http://img516.imageshack.us/img516/5143/warcraftbugky1.th.png] (http://img516.imageshack.us/my.php?image=warcraftbugky1.png)

When I try to execute the following example, I get an error message: /usr/local/lib/ruby/gems/1.8/gems/mocha-0.5.4/lib/mocha/object.rb:40: in `expects'': undefined method `stub'' for nil:NilClass (NoMethodError) from test8.rb:5 What could be the reason? I tried with the latest Mocha Ruby gem, and I also tried it with the Rails plugin. The example: require

I just went through an exercise of writing a Perl script called from my Asterisk dialplan to look at a list of area codes and exchanges to determine which ones are local (no or little cost) under my current Verizon plan. I route calls outside of my local limits to Gizmo. It works fine but when I called Verizon to change (lower) my service it was a bewildering spider web of rates structures just in

I've been trying to use a set-up whereby I have several TA's connected to an Asterisk server (1.2.4) and they act like they're in a hunt-group i.e. try the first, if busy jump to the next etc. in my extensions.conf I had something like [inbound-trunk] exten => 441234123456,1,Dial(SIP/s1a,20,r) exten => 441234123456,102,Dial(SIP/s2a,20,r) exten =>

Hi list, I hope this isn't in error but if it is I apologize. I have a small project request on hand where the clients want their customers to be able to dial in to conduct business over the phone in a completely automated manner. From my limited understanding this looks a lot like a call center where one has to build some sort of proxy that understands their business logic and that can

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex

Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to the US. The IP address of the remote end changed (though in the config file it's registered as a name i.e. asterisk.remote.end), my system didn't recognised the IP change, it must be cached once and then the cached value used for ever. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857

Someone on this list uses the address @sedwards.com I doubt this is their actual email address as there is no MX record for sedwards.com and I can't find registration for their domain either. Part of my mail servers reject these emails because they cannot be replied to, or are likely to be spam. Every so often I get a mail from the list management to say that I've been unsubscribed

Ok, I see there's alot out there of voip providers. Curious what to watch out for ? charges and fee's, etc ? If anyone has feedback as to a GOOD voip provider experience (one that gave FREE DID) Please share. Again, I am doing this to learn about asterisk, I'm currently testing it at home. thanks, On Wed, Mar 17, 2010 at 11:49 PM, Joe Greco <jgreco at ns.sol.net> wrote: