First I have a correct conection with database mysql so the problem is the following:
I want to configure in realtime the sip.conf so:
First into my database I put into schema asterisk and table ast_sipfriends :
id name host type callerid contexto secret username
1 80 dynamic friend 80 internal 80 80
2 81 dynamic friend 81 internal 81 81
And in the sip.conf I have the general context like that:
[general]
context=default
bindaddr=0.0.0.0
bindport = 5060
videosupport=yes
disallow=all
allow=gsm
allow=alaw
allow=ulaw
allow=speex
allow=h264
allow=h261
allow=h263
allow=h263p
language=es
allowsubscribe=yes
asterisk sip allowoverlap=yes
caninvite=no ; These setting confirm we want the PBX handling the audio
canreinvite=no
nat=yes
dtmfmode=rfc2833
jbenable=yes
maxcallbitrate=384
And in my extensions.conf the following:
[internal]
exten => 80,1,Dial(SIP/80,26)
exten => 80,n,Hangup()
exten => 81,1,Dial(SIP/81,26)
exten => 81,n,Hangup()
Into my extconfig.conf :
sippeers => odbc,asterisk,ast_sipfriends
sipusers => odbc,asterisk,ast_sipfriends
The problem is that 80 and 81 are registered but when I try to make a call the asterisk console show:
-- Executing [81@internal:1] Dial("SIP/80-00000000", "SIP/81,26") in new stack
[Jun 11 11:13:57] WARNING[2887][C-00000000]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [81@internal:2] Hangup("SIP/80-00000000", "") in new stack
== Spawn extension (internal, 81, 2) exited non-zero on 'SIP/80-00000000'
And when execute sip show peers 80 and 81 were not in there so this users are registered but can not establish a channel because asterisk does not see them so help me please!