similar to: Phone routing - curious what others are doing?

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

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

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

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?

I just got the list confirmation and noticed it''s text only email so here it is again in plain text. Below is the oringal message. Hi all, I am really struggling with this one, I have built a lot of linux machines using IPSEC tunnels and shorewall gateways. I decied to build a new test machine with Debian running 2.4.25 and Shorewall 2.0.15. I have two subnets on their own switches and

I have a client that wants to put a phone on their web page for customers to call them via their Asterisk server. ) A keypad is needed to enter credit card details. ) "Speed dial" buttons like "Tech Support," "Sales," etc. are a requirement. Actually, passing the SIP address in the HTTP link would work with a bit of arm twisting. ) Free is preferred, but not a

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]

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

Hi there, is gizmo the first user of the Digium Skype solution, or do they use a different approach/product - any clue? http://www.gizmo5.com/pc/opensky/ Philipp

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

I've got an Asterisk server, and several SIP phones behind our router here. Things are working just perfectly inside the network, just as the should. However, I'm not trying to configure my asterisk server to talk with SIP services outside our network, once such example is my gizmo project account. This isn't working out to well. Would it be useful to have a SIP proxy outside of my

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 all, I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a while and it works fine .... I just added CALL OUT ... I have no problem with call setup ... the called party hears me ... but I can't hear them .... again if the call comes INTO the server both sides work fine. Just looking for some tips at where I should be looking .... firewall port forwarding ....

Hi, folks. I've installed icecast 1.3.11 and I'm very happy with it. But I'd like to add a "now playing" gizmo to my home page; can anyone point me to a php script or something else that does the trick? Thanks in advance. -- People don't quit playing because they ___vvz /( grow old. They grow old because they <__,` Z / ( quit playing. [Oliver Wendell

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 been getting lots of failed SIP registrations to a system here. All coming from Taiwan. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Hi All, Until now I've only used IAX2 to connect to ITSPs. I've been toying with a SIP connection to Gizmo Project, but not yet successfully. It brings to mind a question. At what point does it make sense to consider a SIP-aware firewall such as those from Ingate? I'd hate to move away from my m0n0wall, which is open source, easy to manage and has served me brilliantly for two

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I have been using GiZMO for about 6 months now - love it!. You do have to pay for some services, but its nice to have the knowledge that you are not being listened to or your private info is not given out. The only reason that Skype is worth having is because Opera uses it. Now, are you Opera's B-tch?

Is it possible to get Joburg DIDs (probably need 4 at the moment), to be delivered via SIP preferrably to UK. If it's legal, please send pricing. Thanks Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com