Snowulf - VoIP

Dial-Peers on Cisco 3600

nospam@example.com (Jon) — Thu, 25 Jan 2007 14:37:35 -0800

When I setup the Cisco 3600 in the office to work with asterisk as our FXO ports, I needed incoming and outgoing capabilities. In was easy enough, but out was more tricky because you need to setup specific dial-peer plans. Well for the longest time I had 3 different sets of rules per port (911, Local & Longdistance). Luckily I only had 4 ports, for a total of 12 rules. I thought that was stupid, but never got to fixing it. Recently I have. I found the regex for destination-pattern. Here's what why new rule's look like:

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.

NuFone

nospam@example.com (Jon) — Tue, 29 Nov 2005 00:01:00 -0800

NuFone is once again trying to screw me over - this time for 7cents. I dont belive these people, more specifically that one person. I'm preety sure NuFone is just one person. Their service is really bad, and I cancled it after a week of using it last... January. I cancled my account then I asked for a partial refund, wouldnt do it. Filled a paypal complaint, didnt go very far as usual. I figured as long as my account was cancled - I didnt really care any more.

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).

Silly 7960's

nospam@example.com (Jon) — Thu, 16 Jun 2005 18:23:32 -0700

Brought a new 7960 online today in the office - and it paper weighted itself. It loaded up the Universal Application Loader, but then refused to load the new SIP firmware. I found two things that are useful, first Voip-info.org has a useful bit and second - pay very close attention the the TFTP server logs. In the end, copying the .sb2 firmware image to .sbn was the magic trick (and possibly a few extra config files they mentioned on the page above).

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"

When working with Cisco

nospam@example.com (Jon) — Tue, 26 Apr 2005 10:26:16 -0700

If your making changes you want latter - Sugestion: Always write the configuration to memory. We haven't been using our old voip router recently (newer gear for cooler stuff), so I was playing with it a while back (you may remember FXSs are a go! ) - Except I forgot to save my changes to memory before I was done. Sigh. I'm glad I wrote that blog entry, most everything I need to get the system going happily was there.

I spent alot of time fighting with Asterisk yeterday - 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 shiney 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 comming 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.

Sigh and DoubleSigh

nospam@example.com (Jon) — Mon, 14 Mar 2005 09:25:44 -0800

Surfing the RSS feeds this morning and I found two new intresting topics to bitch about in the wide wonderful world (WWW) of technology.

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 320x320 - 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)

Asterisk Goodies

nospam@example.com (Jon) — Sun, 06 Mar 2005 23:50:56 -0800

So this bit was posted on /. . I thought I'd relate the information seeing how its rather nifty guide. Kerry Garrison used Asterisk@Home (which I'll admit I've never heard of before, But I've seen it done similar before) to build a PBX. I think the real key of this installation is that A@H comes with AMP already built in and ready to go. This is a really nice feature seeing is how Asterisk doesn't understand config files. In fact if your a scripter you'd understand the configs better (Because they are indeed micro scripts).

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.

FXS's are a go!

nospam@example.com (Jon) — Mon, 31 Jan 2005 15:24:10 -0800

Well its amazingly difficult to find documentation about how to make FXS's work with Asterisk. I happen to have 4 FXS ports on the Cisco 3640, the same one that has four FXO ports. Below I'm including my configuration for use by others. The FXS port is configured as extention 600 (the Asterisk side configuration is also include).

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 recieve 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 cavet I have is that all the Dialplans 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.

Yea for PBX's

nospam@example.com (Jon) — Sun, 30 Jan 2005 13:41:55 -0800

So several months ago I moved our small office over from a Windows 2000 box running some microsoft built-in IP telephony software, to an Asterisk PBX. Now I'll admit I knew almost nothing about Asterisk, or Cisco hardware. Since then I've made major stides. Yesterdays project was to see if I could get a Cisco 3640 router with 4 FXO ports to connect to Asterisk. I did, and I got it to work.

If your trying to do the same thing, I have a few sugestions (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 sugest 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 explination 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 explanitory, but if your intrested in VoIP on your 3600 series router, take a look at this product guide.

Asterisk!

nospam@example.com (Jon) — Wed, 26 Jan 2005 10:39:33 -0800

So your looking into finding cheap long distance for your IAX2 compatible system (for those that don't know what I'm talking about, I'm talking about VoIP terminator services for your Asterisk PBX). I've been playing with a few of these services recently. I have to say that VoipJet works. They aren't anything fancy, you have to prepay, but their systems work. You can fake your own Called ID, and all those good services.

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.