dial-peer voice 100 pots
destination-pattern …%
port 0/0/0
forward-digits all
!

The above will take all calls that are 3 digits or longer and dial them exactly. (Note: periods are wild cards, and % are 0 or more of the preceding character). The key is to have asterisk send exactly the digits to call and no more. If your like my office, you have an external access number (in my case it is 8). Using ${EXTEN:1} you can drop that in asterisk before it calls out to the cisco. Below is an example of my asterisk’s extensions.conf (Note: 10.10.10.10 is the example IP for the cisco)

[outgoing-calls]
ignorepat => 8
exten => _8011.,1,Dial(SIP/${EXTEN:1}@10.10.10.10)
exten => _81XXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@10.10.10.10)
exten => _8NXXXXXX,1,Dial(SIP/${EXTEN:1}@10.10.10.10)
exten => _8911,1,Dial(SIP/${EXTEN:1}@10.10.10.10)
exten => _8611,1,Dial(SIP/${EXTEN:1}@10.10.10.10)
exten => _8411,1,Dial(SIP/${EXTEN:1}@10.10.10.10)

I could have setup a wild card for 911,611,411 and others, but I’d rather have specific control over what number people can call. Of the same is true with the international calling, but for now I don’t have to worry about that.

Until today – when they send me another bill. Its not like 7cents would kill me, but its the priciple of the thing. First they dont close my account (but I’m not paying for it) then they have the nerve to tell me I owe them money for not using their service. They told me to go online and I could check my account history for myself – guess what…. Nothings happened on my “account” since Jan 20th.

If you can avoid NuFone – I’d sugest it. Its all about the people – and theirs sucks (and so does their disfunctional service).

But I’m rather happy that these 400$ desk ornaments actually have the inteligence to un-paper weight themselves from the dreaded “Protocol Application Invalid”

I spent a lot of time fighting with Asterisk yesterday – So I’ll post a few bits on what I learned a little latter.

Oh – Also, I found that even old FXS’s support two lines in one (what ever the technical term is, I forget – where you have two phone lines in one cable). I currently have a nice shiny 2 line 2.4Ghz portable phone plugged into one FXS jack (2/0/0) but sadly I cant use the other port (2/0/1) also. Would be really nice if I could have 8 lines coming from the 4 ports – Oh well. I guess the FXS is smart enough that if you open the second line on port 0 – It takes from the second port. I configured 2/0/0 with ID of 600 and 2/0/1 of 601 – and it works properly with both caller ID’s.

First off is a moving article from ars technica, about the new Google mobile search. Ars’ first and foremost has a problem picking a side, their poking at it claiming that it could be evil, but its not such a bad idea. Make up your damned minds. Personally I think its a great idea. Now as a website designer I can be lazy and not have a WML compliant version, but still have mobile users that can read my site. While many may argue that this is terrible – More Lazy Webmasters – and all that jazz, I really don’t care about users on cellphones or PDA’s. For the record, yes I do have PDA, in fact I have a Tungsten C with built in WiFi. Note, the screen is 320×320 – If its not specifically designed for a palm screen – it doesnt work. Besides, in my time, I have seen some terrible WML-Versions of sites. So google can probably do a better job, Either that or it will force webmasters to code their sites in good WML so that users will use the sites WML version and not the google proxy version.

Second is another wonderful joy. Military certifies Cisco telephony. Yea! Let us rejoice in knowing that the DOD could use Cisco IP phones – and that they may be good enough for us now. Along with that note I’d like to point out the fact that I have an eMachine (600mhz celeron, with 128mb RAM) in the server closet at work, and its running Asterisk PBX. The machine has an uptime of 45days (we had an extended power outage) and has done a very good job that entire time, and the serveral months before that. So why do products need be certified “that their products are secure and operate without performance degradation in a multivendor environment.”? I’m sorry, my lil ol eMachine runs Asterisk very well – Using Cisco 7960 hardphones, and softphone clients on serveral OS’s alike (Oh, did I mention I used thoes Softphone clients over VPN? ::gasp:: security ::gasp:: reliability ::gasp:: multi-vendor)

Along with my list of misc projects I’ve wanted to do building a PBX at home as been one (Already did it at work). So now this is going to move up on my list, and be even easier.

Cisco 3640 Config
voice-port 0/0/0
station-id name test
station-id number 600
!
dial-peer voice 600 pots
!600 above doesnt "mean" anything
destination-pattern 600
port 0/0/0
!Any call matching 600 goes to port 0/0/0 (my FXS port)
!
! the 2 groups below are calling plans for outgoing calls,
! be it from the FXS or FXO ports
dial-peer voice 402 voip
destination-pattern 2...
!Matches '2' then 3 digits
session protocol sipv2
session target ipv4:192.168.xx.xx:5060
!asterisk server IP
codec g711ulaw
no vad
!
dial-peer voice 401 voip
destination-pattern 7..
!matches 3 digits starting with a 7
session protocol sipv2
session target ipv4:192.168.xx.xx:5060
!Asterisk server IP
codec g711ulaw
no vad
!

Asterisk – Sip.Conf
[600]
type=friend
username=600
host=192.168.xx.xx
; Cisco Router IP
canreinvite=yes
dtmfmode=inband
qualify=1000

Asterisk – Extentions.conf
exten => 600,1,Dial(SIP/600,15,t)

I plugged in a wireless phone and was able to wander around the office calling other IP phones (including my “service” numbers, voicemail and the like, which I have at 2500). I was also able to receive calls. I was rather amused as it worked very very easily. I don’t know if DTMF would work in IVR systems (which I should have checked, but I didn’t). The only caveat I have is that all the Dial-plans I have for the 7960′s would have to be setup on the 3640 (1+10digits, 7+2, 2+3, etc). Note: If you have the router set to match a pattern of 2…. – It wont call through unless you dial 4 digits, it will just hang, much like your normal phones.

If your trying to do the same thing, I have a few suggestions (that were painful to find out). While making one call in is easy, getting key tones to carry over is a different story. I’m putting in the configs I used below, in case it helps anyone else: (Note: I’m using SIP & ULAW codec since its inside a LAN. I suggest that the connection from the Cisco unit be ULAW, while it takes the most bandwidth, it keeps the best quality, and allows for DTMF to work.

Asterisk – SIP.Conf:
[192.168.xx.xx]
type=friend
host=192.168.xx.xx ; IP address of Cisco gateway
dtmfmode=inband
disallow=all
allow=ulaw

Cisco – running-config
voice-port 2/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar 2900
description test input line1
ring number 2
!
dial-peer voice 400 voip
!400 above doesnt actually mean anything
destination-pattern 2900
!match patter from voice port
session protocol sipv2
session target ipv4:192.168.xx.xx:5060
!IP address of asterisk server
dtmf-relay cisco-rtp
!Some how this DTMF works with 'dtmfmode=inline', in asterisk
codec g711ulaw
no vad
!
sip-ua
sip-server ipv4:192.168.xx.xx
!IP Asterisk server
!

Quick explanation above. A call comes into slot 2, VIC0, port0 (While VIC’s are labeled and so are the port numbers, if you need to know how the slot numbering works you can google it or I found this on Cisco’s site for 3640′s) and its sent to peer 2900 on the asterisk server. Peer 2900 can be your IVR (which is what I have) or it can be a normal extension. Most of the configs for the Cisco are self explanatory, but if your interested in VoIP on your 3600 series router, take a look at this product guide.

Now, for a service I recommend you steer clear of, so clear you at least on the other side of the milky way NuFone. Their website is terrible, their “support” department (note, you cant find their email address for support anywhere on their site). Worst of all, they have no idea what’s happening. When you sign up, they don’t send you an email of how to config you system or activate your account (they claim they will, but they done). I stumbled into their web interface which looks like a 5th grader programmed it (I could make a better web interface using QBasic and I’m not a big programmer). I turned on my incoming calls and configured them in Asterisk, worked just fine. Couldn’t make outgoing calls for the life of me. After being insulted by their support team for not using the config settings they sent me in the initial email (note, the email I never got), they sent me new configs. I did not user their info, I tried my service again (this is after 48-72 hours of trying to deal with them) and it worked. Guess what, they never turned on my fragin service in the first place.

Now I have to get PayPal involved (that’s how I paid NuFone, thankfully) to resolve this. I’ve emailed NuPhone’s support department a number of time with no response anymore to canceling my account. It’s sad day, I really hate to do this.