Trip
Sorry, but I suck at explaining stuff :P
Just spent all night trying to get Asterix configured.
I'd just like to say...
my head hurts :blink:
I'd just like to say...
my head hurts :blink:
Mumble - Voip
- Thread starter techrock
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mindlord
Notices Two Things
my head hurts :blink:Tripmonkey_uk said:
I would devote a day to it myself. I have an expert in my office that could help me, but I'm busy with important projects at the moment. It also doesn't help that my home server doesn't have modem installed.
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Trip
Sorry, but I suck at explaining stuff :P
Normally I would but I'm just starting to learn Python so all my spare time going towards that goal at the moment.mindlord said:
I'll have another crack at it later on when time permits.
Mine doesn't have a modem installed either but it should still be useful for VOIP
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mindlord
Notices Two Things
My lappy does, so when I get a few free hours and a hardline, I'll see what I can fiddle with. I've already installed all the packages and have the server running, but it doesn't do anythingTripmonkey_uk said:Normally I would but I'm just starting to learn Python so all my spare time going towards that goal at the moment.
I'll have another crack at it later on when time permits.
Mine doesn't have a modem installed either but it should still be useful for VOIP
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Trip
Sorry, but I suck at explaining stuff :P
Yeah I got it all installed on my server, but zip happened there either.mindlord said:My lappy does, so when I get a few free hours and a hardline, I'll see what I can fiddle with. I've already installed all the packages and have the server running, but it doesn't do anything
Some info that might help?..
sudo nano /etc/asterisk/sip.conf - to change the sip settings.
asterisk -r - to re-start the asterix server after making changes.
You might be best off changing it to work on port 8080 if you already have a web server up and running?
sudo nano /etc/asterisk/http.conf - is supposed to let you configure the built in web interface, but I couldn't get it up on mine for some reason?
http://servername:8080/asterisk/static/config/cfgbasic.html - to access the web interface.
I'll post some more if I find them for you
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mindlord
Notices Two Things
I've actually been able to use twinkle from my lappy to my home PC to do an echo test. I can't tell if my microphone is not working or if asterisk just isn't getting anything from me. It's unlikely that the latency is so low I can't tell the difference between me talking into the mic and the stuff coming out of my headphones. Regardless I can hear the recorded message from asterisk informing me of the echo test before it starts, but it doesn't seem to recognize dtmf tones or anything. I don't think that twinkle is actually transmitting anything at all, or if it is transmitting asterisk isn't getting the stream.
It's a start anyways.
It's a start anyways.
Svartalf
Active Member
my head hurts :blink:Tripmonkey_uk said:
:lol:
I could've told you that this would be what happened. It's a PBX. It's going to have the old evil type of interface that one would expect to find on a PBX or a telecom switch (Honest...).
It's NASTY. I suggest getting the AsteriskNOW installer CD as it'll take MOST of that pain away from you to a nice web-based configuration tool.
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mindlord
Notices Two Things
I have to find a supported dial-up modem to use as the gateway now, but everything seems to be configured properly. Fortunately, I work at an ISP so there are modems laying around everywhere, so it's likely one of the few modems that can be used as a FXO will be around. If not there is a telco switch on the second floor that probably has some back-up PCI FXO's that I could snag cheaply.
javaJake
Jacob Godserv
So, wait a moment. All that's required is a supported dial-up modem and you can use a box at home as a VoIP->Phone "converter"!? That's it!?
You mean folks with Skype or Gizmo are paying through the nose for nothing?
You mean folks with Skype or Gizmo are paying through the nose for nothing?
mindlord
Notices Two Things
Yes, I spent a few hours yesterday talking to the Asterisk expert at my office and he said not only is it possible it's actually quite easy. All you need to do is comfigure the SIP feature of Asterisk (20 or so lines of conf) and configure zapata for the analog line (10 or so lines of conf) and configure an extension so that dialing 9 on the Voip line transfers you to the analog line (10 or so lines of conf).javaJake said:So, wait a moment. All that's required is a supported dial-up modem and you can use a box at home as a VoIP->Phone "converter"!? That's it!?
You mean folks with Skype or Gizmo are paying through the nose for nothing?
Plus you get a full featured voicemail system, call routing system, and all manner of nifty stuff if you take the time to learn how to set them up.
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javaJake
Jacob Godserv
Sounds like a HOWTO-worthy item to me.
One of these Saturdays I'll have to give it a go on an old P3 box I've got sitting here. Too many projects, too little time.
One of these Saturdays I'll have to give it a go on an old P3 box I've got sitting here. Too many projects, too little time.
That sounds like you might be behind a NAT firewall of some sort.mindlord said:
Tim
Yup. You can get the single port FXO cards on EBay for less than $20 now. Search for FXO and Asterisk.javaJake said:So, wait a moment. All that's required is a supported dial-up modem and you can use a box at home as a VoIP->Phone "converter"!? That's it!?
You mean folks with Skype or Gizmo are paying through the nose for nothing?
Still, without the FXO cards, they still make VOIP/VOIP calls just fine. I had set up a bunch of * boxes a couple of years ago for small hotels just to call room to room or the front desk. $80 grandstream phone in each room and a cheapo PC with a 40G drive.
Tim
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