I spent some time (well alot of time) trying to get IPkall to transfer calls to an Asterisk extension. I thought I'd post my config here for others...

IPkall: I entered my IPkall DID as the SIP phone number on the IPkall website, ie.. 360XXXYYYY. Also entered the SIP Proxy of my AAH server, ie...MyServer.org

AMP - Setup - DID Routes: DID Number entered as the IPkall DID number ie.. 360XXXYYYY then selected an extension.

Maintenance - edit configs - sip config: added the following line:
context = ext-did

That's all it needed.

Man, I still get a busy signal every time I dial my ipKall number. Sheesh, I've tried everybody's tutorial too! AAaaaaaaaaaarrrrrrgggggggghhhhhh!

I can't figure out what could be wrong. I've followed the simple instructions. Crud!

Thanks,

-Rob.

I spemt some time trying to get this to work as well. Nerd Vittles
has some OK instructions here, but it didn't work for me. What did work, though, it this:

In the [from-sip-external] sections of your extensions.conf use the following: (it follows nerd vittles instructions)


exten => ipkall,1,dial(IAX2/101,20,m)
exten => ipkall,n,VoiceMail(101@default) ;voicemail
exten => ipkall,n,Hangup

this will send al IPkall numbers to extension 101

or you could include this as the first line in your [from-sip-external] context of extensions.conf to enable DID management:

include => ext-did


The real source of the problem is the following 3 lines in the [from-sip-external]

exten => _.,1,AbsoluteTimeout(15)
exten => _.,2,Congestion
exten => _.,3,Hangup


These lines cause all incoming lines to recieve congestion and never make it into your asterisk server. Make sure anything you add to your [from-sip-external] context is BEFORE these lines and it should work. I dicovered this little problem by using CLI to debug my problem with IPKall.

Rob-

If you still get a busy signal then try waiting 24 hours. I've gotten a busy signal after configuring IPKALL then it starts working the next day. Also, doubble check your domain name is pointed at your correct IP address with a ping command. This is all I do to get it working.

Note: On AAH 2.0 The "DID Routes" is now called "Inbound Routing" and I had to leave "Caller ID Number" blank to get it to work.

Here is my sip.confg from AAH. Don't forget to click "update" to save the config file and then click "Re-read Configs" at the top of the page to reload the sip stack.


[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
context = ext-did

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

Thanks, I'll give it a try. I really appreciate your help!

Oh, by the way, if you have your ipKall number pointed to a ring group or auto attendant on your * box, and say one person calls and is interacting with the menu or talking to someone in the ring group that answered and a second, third, etc. person calls, will those subsequent callers be able to connect via the same ipKall # assuming you have enough bandwidth? I'm wondering if the ipKall just kind of transfers and releases to your asterisk, wondering if callers to your ipKall number would ever get a busy signal assuming your asterisk still had an available line and or bandwidth. Do you know what I mean?

Thanks again,

Rob.

Rob,

I just tested using my cel phones and was able to call the same ipkall number 3 times - so six phones and and 3 independent conversations using ipkall asterisk and ring groups. My upstream is only around 280k but all calls were clear. (note: this test was limited by the number of phones I had laying around!)

It would be nice if ipkall would allow a reinvite to a remote extension from my asterisk box. This would cut down on bandwith on the asterisk node.

The reinvite works for sipphone.com when my remote extension has a public IP address. I just configure canreinvite = yes for that extension.

I'm using shorewall on my asterisk box for routing and am very happy with it.

Tom