RE: [asterisk-users] Refresher course needed!

2007-03-30 Thread Yuan LIU

From: "Brad Sumrall" <[EMAIL PROTECTED]>
Date: Tue, 27 Mar 2007 00:06:13 -0500

Hello everyone

My name is Brad, I am an old Asterisk Vet of the very early days just 
coming

back to join the group.

Ok, for starters, I feel like the "monkey with the light bulb" looking at
extensions.conf and sip.conf.

It has been some time.

A friend ask me to set up a asterisk server that records phone calls.

FC4
Asterisk 1.4
And all the latest and greatest


Problem number 1

Some good "get back into the grove" literature.
"I work CLI only", never much for graphics and gui's


Asterisk 1.4 still has CLI.  I don't think many people here use GUI.  
voip-info.org is a good starter.  Another really good restarter?  CLI> help!



Problem number 2

We have asterisk logged into teliax but cannot see the inbound call come up
on the CLI

Tethereal says this;
1660   3.829799 207.174.202.4 -> 66.109.17.92 SIP Status: 100 Trying(1
bindings)
1661   3.831357 207.174.202.4 -> 66.109.17.92 SIP Status: 200 OK(1
bindings)

Asterisk says this;
*CLI>

Nothing, notta!


How did you start Asterisk or remote console?  Have you tried "core set 
verbose 10"? (Just kidding.  Most often I go 3.)  Have you tried 
"sip set debug"?



My extensions.conf
(yes, I loaded the samples)
 [general]
static=yes
writeprotect=no
clearglobalvars=no
;#include "filename.conf"

[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest   ; IAXtel username/password
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1  ; MSD digits to strip
(usually 1 or 0)

;From here is brads stuff
exten => _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten => YOURNUMBER,1,Answer()
exten => YOURNUMBER,1,DIAL(SIP/user,20)


Getting more confused about what inbound call you did not see after reading 
the sample conf.  Did you put a context title before brads stuff?  What is 
your sip.conf/user.conf if you expect incoming call from SIP?


Ah.  Feels good to teach grandma cook milk:-)

Yuan Liu


Thanks to all!

Brad



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[asterisk-users] Refresher course needed!

2007-03-26 Thread Brad Sumrall
Hello everyone

My name is Brad, I am an old Asterisk Vet of the very early days just coming
back to join the group.

Ok, for starters, I feel like the "monkey with the light bulb" looking at
extensions.conf and sip.conf.

It has been some time.

A friend ask me to set up a asterisk server that records phone calls.

FC4
Asterisk 1.4
And all the latest and greatest


Problem number 1

Some good "get back into the grove" literature.
"I work CLI only", never much for graphics and gui's

Problem number 2

We have asterisk logged into teliax but cannot see the inbound call come up
on the CLI

Tethereal says this;
1660   3.829799 207.174.202.4 -> 66.109.17.92 SIP Status: 100 Trying(1
bindings)
1661   3.831357 207.174.202.4 -> 66.109.17.92 SIP Status: 200 OK(1
bindings)

Asterisk says this;
*CLI>

Nothing, notta!

My extensions.conf
(yes, I loaded the samples)
 [general]
static=yes
writeprotect=no
clearglobalvars=no
;#include "filename.conf"

[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest   ; IAXtel username/password
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1  ; MSD digits to strip
(usually 1 or 0)


;From here is brads stuff
exten => _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten => YOURNUMBER,1,Answer()
exten => YOURNUMBER,1,DIAL(SIP/user,20)


Thanks to all!

Brad


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