similar to: How to configure Asterisk to handle multiple Gizmo accounts?

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 there everyone! I use asterisk as a home pbx. My internet connection is a DSL one, and I have a Linksys WRT54G that nat things for me in a 192.168.X.X style network. I've installed asterisk on my mac, and tried several examples I've found on the net (voip-info, gizmo, etc.) about how to create a Gizmo and a FWD trunk. However, all my attempts failed. The FWD thing kinda

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

Hello there, I'm a new Asterisk user and I'm having difficulties to connect to and from my Asterisk server. Can anybody give me a hand? Here's some background information: * I'm running RedHat Linux Enterprise 4.0 * When iptables is stopped, my server can register with IAX service providers and receive registrations from IAX softphones. However, it does not succeed in

Hello there, I'm a new Asterisk user and I wonder if it is possible to associate a voicemail box with a group of users, i.e., a single recorded message is sent to everyone in that group. If so, where can I find more information about that? Thanks in advance, Leo Burd

Hello everyone, I'm having all sorts of problems with my PHP AGI scripts... Basically, my scripts run fine from the command line and don't do anything well called from Asterisk. Here are my questions: a) Does Asterisk require PHP CLI or CGI? From the command line, my script seems to work fine with PHP 4.3.11 (cli) but not with PHP 4.3.9 (cgi) b) How to debug my script? According

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 ....

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?

Hi All, At the usual time, 12 Noon ET on Friday April 3rd, we expect Michael Robertson to join the discussion to filed questions about OpenSky and Gizmo5. I have been testing all of these Skype to X methods except SIP for Skype since I have no word from them. I can tell you that we've had good results with bith Skype for Asterisk and OpenSky. In fact, I am currently accepting calls to my

Hi, Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a lot of experience in the telecommunications space and he joins us today to chat about its current state, conferencing and whatever else comes to mind. So we have a meta conference aout conferencing, it won't be the first time :) You probably saw John Todd's message on one of the lists: Skype for Asterisk is in open

Hi, Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a lot of experience in the telecommunications space and he joins us today to chat about its current state, conferencing and whatever else comes to mind. So we have a meta conference aout conferencing, it won't be the first time :) You probably saw John Todd's message on one of the lists: Skype for Asterisk is in open

Hello there, I'm trying to figure out how to run a PHP script on a remote machine and still have access to the audio stream associated with the call. Ideally, I'd love to play/record audio files directly from/to the remote server without having to copy them back and forth to the Asterisk server. What is the best way to do this? Is it possible to combine EAGI with FastAGI in PHP?

Hello there, Somehow, Asterisk log files are consuming all the space that I have in my hard disk... They've already eaten 14GB and are still hungry!! What shall I do? I'm not even logging anything in verbose mode!! Help really appreciated!! Best, Leo

Hello there, I'm new to Asterisk and I'm trying to figure out a way to make the Asterisk Manager Interface (AMI) accessible to multiple users at the same time. Would anyone recommend an AMI proxy that could be accessed from PHP code? Thanks in advance, Leo

Hello everyone, Where can I find instructions on how to install PHPAGI? BTW, what's the difference between PHPAGI and PHPAGI2? Are they different products? It's hard to tell from voip-info.org... Best, Leo

Dear Moises Silva, Chris Thompson and others, I've finally managed to find the problem with my PHP AGI script! Apparently, my machine has 2 versions of PHP installed. One for CGI and the other for CLI. Without knowing, to hand test my script I was calling the CLI version (/usr/local/bin/php), whereas Asterisk was calling the CGI version (/usr/bin/php). It turns out that my script

Hello there, I'm new to Asterisk and I'm wonder what's the best way to combine multiple VOIP lines into a single phone number... I currently have 4 incoming VOIP lines, each one with a different number. Ideally, I would love to provide my customers with a single number for all those lines. Is that possible? If so, would anyone give me directions or send me a dialplan example

Hi Leo, here's a suggestion: in your dialplan (extensions.conf) send multiple users to the same mailbox (e.g. 999) if they do not pick up within 30 seconds: ; SIP Phone 100, Tom exten => 100,1,Dial(SIP/100,30) exten => 100,2,VoiceMail(999) ; SIP Phone 200, Eric exten => 200,1,Dial(SIP/200,30) exten => 200,2,VoiceMail(999) and in your voicemail.conf you do not specify a single

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Anyone got any thoughts on this and how it compares to the chan_skype that's due soon ? "OpenSky is a free service provided by Gizmo5 which allows *any* mobile phone, web browser or IP aware phone network (SIP, asterisk, etc) to communicate with Skype users. OpenSky supports sending text messages and voice calls." http://www.gizmo5.com/pc/opensky/ Julian