MIGRANES due to asterisk

Asterisk Asterisk Support
1

Hey y’all,

I have no clue what I am doing wrong. I installed asterisknow and went through the gui setup. I went in and edited the sip.conf and extensions.conf files for a sip user. I was having a problem when I would call in it would tell me that the user for extension XXXX is not available. I went into the users.conf file and edited the host=dynamic string under [6000](the extension setup for user) and changed it to host=(ip address of phone). Now I can call in and the phone rings when I dial the extension. Should I have to hardcode the phones IP address like that? Another problem I am having is that when I pick up the handset on my Grandstream GPX-2000 and dial 8500 for voice mail I get a 404 not found error as well as when I try to dial out I get a 404 error. I am lost and have been unable to find my way out of this hole. Can anyone please help?

2

If the phone registers to Asterisk then there is no need to specify the IP in the host field. As far as dialing out or dialing 8500, it sounds like they aren’t in the context that your phone belongs to.

3

Well I don’t know what I am doing wrong here. Quite frankly I am getting agitated. I started off this morning with a fresh install of AsteriskNOW and I set the phones back to factory defaults. Here is what I did:

sip.conf
[Bill Hughes]
type=friend
context=test
host=dynamic
dtmfmode=RFC2833
canreinvite=yes
username=bill
secret=hughes

extensions.conf

[test]
exten => 6000,1,Dial(SIP/bill) <-- (is probably wrong?)

voicemail.conf
userscontext=test

users.conf

[6000]
fullname = Bill Hughes
secret = 2187
email = bill@mycompany.com
cid_number = 6000
zapchan =
context = test
hasvoicemail = yes
hasdirectory = yes
hassip = yes
hasiax = no
hasmanager = no
callwaiting = yes
threewaycalling = yes
mailbox = 6000
hasagent = no
group =
host = dynamic

Can someone guide me and let me know what I have done wrong? It all seems like it should be so simple and others I have talked to about it tell me its just a matter if install, setup phone, setup user and viola… But just like all my other open source endeavors its just not as easy for me.

4

Change “[Bill Hughes]” to “[6000]” and change “Dial(SIP/bill)” to “(SIP/6000)”. Also remove the username.

5

I will try that thanks

6

I can call into the phone, but still when I try to dial to check voicemail or try to dial out I still get a 404 not found error. Thanks so much for your help so far though. Hopefully I will have a grasp of asterisk before deployment in the real world.