Basic setup for internal calls

Asterisk Asterisk Support
1

Hello experts,

I am trying to setup 2 480i (extn 2001 and 2002) to call each other internally but so far is no where near success. I am @ best able to dial extn 2001/2002 and be diverted to the voicemail but other than that, nothin is working.

I had a look from the CLI and it seems that whenever I dial the extensions,
The following error appears :

  1. chan_sip.c:1675 create_addr:No such host: 200x
  2. app_dial.c:959 dial_exec_full:Unable to create channel of type ‘SIP’ (cause 3)
  3. file.c:561 ast_readaudio_callback: Failed to write frame

So far I have only configured the sip.conf and extensions.conf . Can someone kindly advise is there anything else that I must do to get the 2 phones talking ? Thank you very much.

from
Kwok Wei

2

so post them here, 'cos that seems to be what’s wrong.

3

hello baconbuttie,

Heres the conf files. I downloaded these sets of conf files somewhere in the Net and made some minor adjustments to it. If you can spot where I may have gone wrong, kindly do let me know. Thanks !

extensions.conf[u]
[general]
static=yes
writeprotect=no
srvlookup=yes

[home]
exten => _200x,1,Answer
exten => _200x,2,Dial(SIP/200x,10)
exten => _200x,3,Voicemail(u${EXTEN})
exten => _200x,4,Hangup
exten => _200x,103,Voicemail(b${EXTEN})
exten => _200x,104,Hangup

exten => _1nxxnxxxxxx,1,Dial(SIP/${EXTEN}@outbound)
exten => _1nxnxxxxxxx,2,Hangup

;Number that the IP Phones dial to access voice mail
exten => 22999,1,VoiceMailMain (${CALLERIDNUM})

;VoiceMailMai9,s100
exten => 22999,2,Wait(3)
exten => 22999,3,Hangup

sip.conf[/u]
[general]
port=5060
bindaddr=0.0.0.0
context=home
dtmfmode=rfc2833
disallow=all
allow = gsm
allow=ulaw
language=en
host=10.33.158.92

[2000] ;Notebook - X-Lite client
type=friend
username=2000
secret=123456
host=dynamic
dtmfmode=rfc2833
context=home
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
disallow=all
allow=gsm
allow=alaw
allow=ulaw
callerid=Notebook
mailbox=2000@local

[2001] ;Aastra 1
type=friend
username=2001
secret=123456
qualify=1000
;qualify=yes
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=alaw
allow=ulaw
context=home
canreinvite=no
callerid=Aastra 1
mailbox=2001@local

[2002] ;Aastra 2
type=friend
username=2002
secret=123456
qualify=1000
;qualify=yes
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=alaw
allow=ulaw
context=home
callerid=Aastra 2
canreinvite=no
mailbox=2002@local

[2003] ; X-Lite (PC)
type=friend
username=2003
secret=123456
host=dynamic
dtmfmode=rfc2833
context=home
callerid=PingTel
canreinvite=no
mailbox=2003@local

[2004]
type=friend
username=2004
secret=123456
host=dynamic
dtmfmode=rfc2833
context=home
canreinvite=no
mailbox=2004@local

4

that lines a killer … try exten => _200X,2,Dial(SIP/${EXTEN},10) instead

5

Baconbuttie,

THANK YOU VERY MUCH !!!
I apologise for going to ask such a newbie question but can you explain to me why your code works instead ?

6

the ‘X’ in the extension is a numeric wildcard. the Dial() application wants a specific destination, not a wildcard. using the ‘_’ assigns the following digits to the ${EXTEN} channel variable … and Dial() can use that.

if you haven’t done so already, read the sticky at the top of the forum and download and digest the free Asterisk book.

7

Thanks. I just found that out while waiting for your reply. Looks like I have alot of reading to do now rather than to cut and paste from the net. You have been a great help.

Btw, I am still unable to access the voicemail presumely @ 22999. Will you be able to know why is this so ?

8

do you have mailboxes created for the extensions ?

9

Yes I do. Infact, I can see the messages recorded under their respective folders. Problem is whenever I dial the voicemail pilot number @ 22999, an engage tone is always presented.

10

Instead of this…

Try this…

It may be the space between the “VoiceMailMain” and the “(”. The Caller ID variables should both work, but I believe the ${CALLERIDNUM} is deprecated.

And welcome to Asterisk! :smiley:

Yours,
Mike

11

Thanks Mike and Bacon !!
I can access my voicemail now ! There are some erros appearing on the CLI though

  1. pbx.c:1163 ast_func_read: Function CALLERID not registered

Do you know what this means ? Do i need to pay attention to this error ?

from
Kwok Wei