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Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

...would happen is it would simply play the message and wait for the person to press a button, if nothing is pressed, it just keeps going down the list. Any other suggestions? asterisk-users-request@lists.digium.com wrote: >Date: Wed, 15 Jun 2005 00:53:14 -0500 >From: Jon Gabrielson <jon@directfreight.com> >Subject: Re: [Asterisk-Users] Newbie question about pressing a key to > be connected to the caller >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <200506150053.14408.jon@directfreight.com> >Content...

...uld happen is it would simply play the message and wait for the person to press a button, if nothing is pressed, it just keeps going down the list. Any other suggestions? asterisk-users-request@lists.digium.com wrote: > Date: Wed, 15 Jun 2005 00:53:14 -0500 > From: Jon Gabrielson <jon@directfreight.com> > Subject: Re: [Asterisk-Users] Newbie question about pressing a key to > be connected to the caller > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <200506150053.14408.jon@directfreight.com&g...

by "GUI" do you mean a configuration utility or a User Interface? MATT--- -----Original Message----- From: Jon Gabrielson [mailto:jon@directfreight.com] Sent: Thursday, February 10, 2005 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] asterisk GUI's that supports zap fxs extensions Are there any gui's that support zap fxs extensions? AMP seems to be one of the more popular gui's but...

hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: hanksmith4@earthlink.net gmail: hanksmith5@gmail.com msn messenger: hanksmith4@earthlink.net aim: hanksmith5 skype: hanksmith5 -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an adit 600 with an fxo card connected to a digium T1 card. If I try to make an outgoing call and the T1 cable is disconnected, asterisks returns congested like it should. But, if the adit 600 is connected to the T1 card, the adit 600 immediately "answers" the call even if there are no physical lines attached. I even removed the fxo card and the adit 600 still

I have an adit 600 connected to a normal analog line. When I try to call that line, the phone rings a quarter ring(almost a beep) instead of a complete ring and keeps ringing and ringing with asterisk never picking up the call. Outgoing calls on those same lines aren't working either. Any suggestions on what might be wrong? Thanks, Jon.

I have asterisk connected to an adit 600 with fxo ports. When I place a call with asterisk, asterisk will try to dial out on the first line even if the first line is already being used by someone else. Any ideas on what I'm doing wrong? Thanks, Jon.

Are there any gui's that support zap fxs extensions? AMP seems to be one of the more popular gui's but it doesn't support zap fxs devices. Thanks, Jon.

What cvs versions are people using in a production environment? Is there a good way to choose a cvs version, like maybe one right before a major change? Are there any plans on having a "testing" branch that is not as stable as stable, but not as bleeding edge as head? I'm thinking of compiling a list of specific cvs versions that are being successfully deployed, would this be

The persistentmembers=yes is suppose to keep agents in a queue over a restart. It might do this, but it doesn't do much good as even if they all remain in the queue, they are all logged out on a restart. Is there any way to keep the agents that are logged in, logged in across a restart? Thanks, Jon.

I'm trying to get canreinvite=yes to work. I would like asterisk to release the line and let the 2 ports on the sipura device to talk to each other directly. Is there a setting I need to activate on the sipura device, or is there something else I need to do? It's possible that it is a nat problem as the sip device is behind a firewall, but it works fine otherwise. Any suggestions?

I am using the following command and the excluded directory /usr/share/doc/ still appears with all it's contents in the backup directory when it's contents should be deleted. Is this a bug or am I doing something wrong? rsync --delete --delete-excluded -a -R --exclude="/proc/*" --exclude="/dev/*" --exclude="/usr/share/doc/*" --exclude="/backup*"

Ok, this is probably a stupid question, but I can't seem to find anywhere where it tells how to put someone on hold on a zap channel. Flash gives me a dialtone and # tells me to enter a new extension, how can i just put the caller on hold. Pressing # then hanging up drops the call. Is there a simple way of doing this without transfering the user to a parking lot? Thanks, Jon.

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

I am trying to get remote extensions to work correctly with agents. I have ackcall=yes and have agents logged in to extension 101 using agentcallbacklogin with extension 101 defined as: exten => 101,1,Dial(Zap/3/18165551234,20,tTA(custom/presspoundtoanswer)) This setup works great on local and/or voip channels, but on zap channels, the zap channel answers immediately as soon as it goes off