Getting Started -- best approach?

Asterisk Asterisk General
1

I’d like to get started with asterisk. I have several voip phones, many analog phones, and an analog land line. (Well, actually, it’s vonage – but as far as asterisk goes, it’s a land line :smile:

So, my plan is to get a FXS and FXO. I’d prefer ethernet, so I don’t have to crack open my server . . but . . . seems like PCI cards are much cheaper.

So – onto the questions . . .

[ol]
[li]What is a good, inexpensive FXS / FXO solution? (Or – is there some super sekret way to make things work with vonage)?
[/li]
[li]If I wanted to get started screwing around before I had the FXS/FXO, is there anything fun I could do with the asterisk software and a few Cisco IP Phones?
[/li]
[li]I do have a locked Cisco ATA186 that I’m trying to brute force . . . assuming I do manage to find the password, is it even worth messing with? Since, I think it’s only a FXO, and they’re pretty cheap.
[/li]
[li]And, finally, assuming I fall in love with asterisk, is there any way to transfer my vonage number over to a voip provider? And, what is the going rate for the providers? Are any cheaper than $25ish for unlimited US & Canada?[/li][/ol]

Thanks in advance!

2

Cheaper that what…? Be aware that you get what you pay for. The very cheap PCI cards may disappoint you, they will have no echo cancellation and may be difficult to get working.

I believe that to use Vonage with SIP, i.e. Asterisk, you have to sign up for their ‘softphone’, which costs an additional $10 /month.

My above comments not withstanding, I have recently aquired a USB fxo (http://wiki.sangoma.com/sangoma-wanpipe-usbfxo) to connect an Asterisk box to the PSTN. So far, it seems to work quite well. There is no echo cancellation, but the software echo cancellation is working pretty well.

[quote]If I wanted to get started screwing around before I had the FXS/FXO, is there anything fun I could do with the asterisk software and a few Cisco IP Phones? [/quote]Of course, you could call yourself to your heart’s content. Or give a phone to someone else and call them.:laughing:

[quote]I do have a locked Cisco ATA186 that I’m trying to brute force . . . assuming I do manage to find the password, is it even worth messing with? Since, I think it’s only a FXO, and they’re pretty cheap. [/quote] I’m not familiar with that one, but I guess it depends how easy it is to find the password…

[quote]And, finally, assuming I fall in love with asterisk, is there any way to transfer my vonage number over to a voip provider? And, what is the going rate for the providers? Are any cheaper than $25ish for unlimited US & Canada? [/quote]Most DID providers offer number transfer if you sign up with them, although how easy this is from Vonage, I don’t know. The deals all seem to work out about the same (no surprise there.) I use 2 providers (well actually 3, but one’s in France.) Viatalk (http://www.viatalk.com) is $199/year for unlimited US/Canada, and they have been pretty good, I’ve been using them for about a year now. I have also been using Broadvoice http://www.broadvoice.com - slightly more expensive but more flexible deals.

Have fun!
Ian

3

OK, I have made a LOT of progress. I got a couple of Cisco IP Phones loaded and booting to sip. The 7960 seems stable. The 7961 is flaky.

I also added a soft phone line to vonage, and I think I have that extension working properly. So, asterisk sees two extensions (vonage & ip phone) and all seems happy.

And, I actually managed to get a call from the ip phone to go to my cell phone – I heard ringback. But, when I answered, there was no audio, and the call quickly disconnected.

So – no my questions are way more noob-ish. And, before I ask what’s wrong with my configuration files, I thought I’d take a step back, describe what I have, what I want, and make sure asterisk can get me there.

I have 3-4 Cisco IP phones. I’d like to buy a couple of cordless 802.11 phones too. And, I have an existing slew of analog phones. If everything ends up working perfectly, then I’ll use an ATA to get the existing phones on asterisk. But, for now, my testing will be limited to the vonage soft phone.

So – I’d like extensions to be able to do normal extension things. Transfer calls, intercom, and make outside calls via vonage. I would also like them to all ring when a call comes in on vonage. That is what I’m aiming for today.

Soon after, if I still like asterisk, I want to add the ATA. And, I’d like to keep the soft phone for a bit, and have two outside lines. Most phones would have the ATA line as their primary line. But, would still be able to access the soft phone (“business”) line as a secondary. The phone on my desk would be reversed. I know the phones support lots of lines, so this should work fine.

And, finally, my real office is going to be installing a cisco Call Manager soon. I can only assume that asterisk would not be able to hook up with that. Or am I wrong? Having one phone on my desk that could talk to Vonage soft phone, my ATA analog line, and a remote Cisco Call Manager would be awesome. Without this, I need to keep two phones on my desk – which is much less awesome :confused:

Thanks in advance!

-Dave

4

Connection but no audio is typically a firewall issue. Make sure your firewall/router is directing rtp traffic to Asterisk. If that’s not the problem, you may have to configure sip phones on external networks to use a STUN server.

Everything you describe is certainly possible in Asterisk. My experience with Wi-Fi phones is that they really need a strong signal. My Nokia N85 works well as a Wi-Fi sip phone, my Linksys WIP330 less so.

I have no experience with Call Manager, but it looks possible: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration

Good luck!
Ian

5

Ok – RTP / Firewall issues . . . should I just forward a block of ports? I know the RTP port generation is automagic.

And, for the rest of the tweaking / setup, where should I start? Is there a quick RTFM I can read for setting up lines / call routing, etc?

Also, what about key delay – one thing I did notice was the long delay waiting for more keystrokes. I think I’d prefer a dial 9 scenario for an outside line . . or . . maybe some other method to recognize internal extensions versus outside pstn numbers . . .

-Dave

6

The RTP port range is 10000-20000 by default, but you can change this in /etc/asterisk/rtp.conf. As for the rest, if you haven’t done so already, you should start by reading the book: http://downloads.oreilly.com/books/9780596510480.pdf

By the way, I forgot to mention that using an iPhone as a SIP client also works pretty well. The only problem is that iPhone applications can’t run in the background, so you have to have the sip client running all the time if you want to receive calls.

Ian