I have been trying to configure a local asterisk connection to broadvoice via the BYOD plan, but I can’t seem to get it working.
sip.conf:
[general]
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
pedantic=no ; Enable slow, pedantic checking for Pingtel
videosupport=yes ; Turn on support for SIP video
recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;Broadvoice config
register => 2036489552@sip.broadvoice.com:[secret]:2036489552@sip.broadvoice.com/5150
[mysjphone]
type=friend
host=dynamic
dtmfmode=inband
username=mysjphone
secret=testing
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=2036489552
secret=[secret]
username=2036489552
insecure=very
context=from-broadvoice
authname=2036489552
dtmfmode=inband
dtmf=inband
;canreinvite=noextensions.conf
[code]
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command ‘save dialplan’ too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk’s best guess (strongly recommended).
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
autofallthrough=yes
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on an extensions reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
;
clearglobalvars=no
;
; If priorityjumping is set to ‘yes’, then applications that support
; ‘jumping’ to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a ‘j’ option in their arguments.
;
priorityjumping=no
;
; You can include other config files, use the #include command
; (without the ‘;’). Note that this is different from the “include” command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include “filename.conf”
; The “Globals” category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
; variables,
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
;exten => 5150,1,dial(SIP/mysjphone)
;exten => mysjphone,1,goto(5150,1) ; To be able to dial with text, “mysjphone”
exten => _1NXXNXXXXXX,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX,2,congestion()
exten => _1NXXNXXXXXX,102,busy()
;exten=_01130.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01131.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01132.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01133.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01134.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_011351.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_011352.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_011353.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_011378.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01139.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01141.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_011420.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01143.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01144.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01145.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01146.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01147.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01148.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01149[2-9].,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01154.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01155.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01156.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01160.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01161.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01164.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01165.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01181.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01182.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_011852.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_01186.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_011886.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_011972.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
;exten=_011.,2,congestion() ; No answer, nothing
;exten=_011.,102,busy() ; Busy[/code]
Ever time I start up asterisk I get this error:
Asterisk Ready.
– Got SIP response 409 “Conflict” back from 147.135.20.128
Please any help you could give me would very helpful. I think I am going out of my mind since all the instructions seem to make it look like I am the only person in the world who can’t get this to work.
Thanks in advance.