[asterisk-users] setting up phones

[Danny Nicholas]

What do you get from "sip show peers" in CLI?  Do you have your ip address
in sip.conf?

 

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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ott Rose
Sent: Thursday, July 09, 2009 4:12 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] setting up phones

 

I followed it the best I could. the phones say no service. I haven't got to
setting up the SIP trunk yet I was told I could get the extensions to work
so I could test between the two phones i have. I have to nics in my server.
one is connect to the phone router the other to a network switch. which ip
should it point to? I am guess the one connected to the switch. That is the
one i can access the GUI from. Below are my users.conf setting. Notice all
the spaces. I didn't put them in there they are like that in the conf
[501]
username = 501
transfer = yes


mailbox = 501
call-limit = 100

type = peer
fullname = 501

registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 501
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
secret = 501
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
disallow = all
allow = ulaw,gsm
macaddress = 00085d10927f
autoprov = yes
label = 501
linenumber = 1
LINEKEYS = 1


[500]
username = 500
transfer = yes


mailbox = 500
call-limit = 100

type = peer
fullname = 500


registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 500
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
secret = 500
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
macaddress = 00085d1095aa
autoprov = yes
label = 500
linenumber = 1
LINEKEYS = 1

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From: danny at debsinc.com
To: asterisk-users at lists.digium.com
Date: Thu, 9 Jul 2009 14:03:50 -0500
Subject: Re: [asterisk-users] setting up phones

It should be pretty simple.  Follow the instructions on this page

http://www.voiptalk.org/products/aastra-setup.html

put the username from sip.conf into the first 4 fields, the secret into the
password field and your asterisk ip into the fields that say voiptalk.org

 

users.conf

[207]

username=207

transfer=yes

mailbox=207

call-limit=2

fullname=mickey mouse

registersip=no

host=dynamic

callgroup=1

context=default

cid_number=207

hasvoicemail=yes

vmsecret=1234

email=user at yourpbx.com

threewaycalling=yes

hasdirectory=yes

callwaiting=yes

hasmanager=yes

managerread=system,call,log,verbose,command,agent,user,config

managerwrite=system,call,log,verbose,command,agent,user,config

hasagent=yes

hassip=yes

hasiax=no

secret=xxxxx

nat=yes

canreinvite=no

dtmfmode=rfc2833

insecure=no

pickupgroup=1

macaddress=00000000001170

autoprov=yes

label=207

linenumber=1

disallow=all

allow=ulaw,gsm

 

sip.conf

[207]

type=peer

context=phones

host=dynamic

fromuser=207

call-limit=3

secret=xxxxx

canreinvite=yes

directrtpsetup=no

nat=yes

qualify=yes

register => 207:xxxxx at yourpbx.com/207

defaultip=1.2.3.4

mailbox=207

disallow=all

allow=alaw

 

 

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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ott Rose
Sent: Thursday, July 09, 2009 1:52 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] setting up phones

 

Can someone tell me how to setup a Aastra 75i phone? I have been trying to
set it up and have pointed it to our asterisk server and selected http for
download. What is the path? I have created two extension in asterisk for
testing. I can't even get the phones to call each other.

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