[Asterisk-Users] extension.conf and SIP phones.

Ernest W. Lessenger ernest at oacys.com
Mon Sep 8 12:37:27 MST 2003


At 03:17 PM 9/8/2003 -0400, you wrote:
>We would like to setup in house SIP phones with numbered extensions for 
>demonstration purposes.
>
>What is the syntax to associate a extension with SIP phone?

exten => 1234,1,Dial(SIP/username)

>Does the Dial application have a SIP specific entry for example: 
>Dial,SIP/SIPphone/s|15

See above.

>When I call from one extension to another I get "User is on the phone".

The SIP phone needs to have logged-in in order for you to dial their 
extension. The usual reason for not being able to dial a SIP phone is that 
(a) they did not actually log-in correctly, or (b) you've rebooted asterisk 
since they last logged in.

>sip.conf
>[user]
>callerid="User Name"
>type=friend
>nat=no
>username=user
>secret=password
>host=dynamic
>canreinvite=no   ; Cisco poops on reinvite sometimes
>qualify=200   ; Qualify peer is no more than 200ms away
>defaultip=10.1.1.53
>dtmfmode=rfc2833
>mailbox=1115
>
>voicemail.conf
>1115 => 1234,User Name,user at somecomapany.com
>
>extension.conf
>exten => 1115,Macro(stdexten,1115,${user})
>exten => 1115,1,Dial,SIP/user/s|15
>exten => 1115,2,Voicemail,u1115
>exten => 1115,102,Voicemail,b1115
>exten => user,1,Goto(1115|1)

Try this:

sip.conf
[user]
callerid="User Name"
type=friend
nat=no
username=user
secret=password
host=dynamic
canreinvite=no   ; Cisco poops on reinvite sometimes
qualify=200   ; Qualify peer is no more than 200ms away
defaultip=10.1.1.53
dtmfmode=rfc2833
mailbox=1115
context=default

extension.conf
[default]
exten => 1115,1,Macro(stdexten,1115,SIP/user)

[macro-stdexten]
exten => s,1,Dial(${ARG2},20,tT)
exten => s,2,Voicemail2(u${ARG1})
exten => s,3,Goto(default,s,1)
exten => s,102,Voicemail2(b${ARG1})
exten => s,103,Goto(default,s,1)




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