Asterisk settings Allo.com

Asterisk Configuration Examples


Incoming and Outgoing Settings



here's a basic asterisk configuration to place and receive calls through allo.com network.

for the sake of simplicity, lets replace your sip.conf file by the following (pls make a backup!!):


[general]
context=default ; default context
srvlookup=no ; sip srv lookup applies to allonetwork.net, and not to specific proxy IPs
allowguest=no ; disallow unidentified sip calls to come in, useful for debugging

NOTE
dont use externip for NATings since its taken care by proxy itself.

  1. include sip_additional.conf


then lets create sip_additional.conf file, in which we'll keep the peer and registration configs:


register = username:password@67.17.211.246/username
; fastest (and closes) proxy to me is in NYC (get proxies on www.allo.com/asterisk/)

;Allo.ComTrunk Setting
[allo.com]
type=peer
username=username
fromuser=username
secret=password
context=from-allo
host=67.17.211.246
fromdomain=allo.com
insecure=very
disallow=all
canreinvite=no
dtmfmode=inband
allow=ulaw
allow=alaw


now, lets create a context for the incoming and outgoing calls of this peer, in extensions.conf



; Note: Make sure you use the '/username' at the end of register line in your sip configuration.

[from-allo]
exten => username,1,Goto(default,s,1) ; assuming you have a default context where incoming calls are routed

;or use the following dialplan to route based on the DID's

[from-allo]

exten => username,1,System(echo \'${SIP_HEADER(TO)}\' | egrep "sip:.*5555555555@" > /dev/null)
exten => username,n,GotoIf($[ "${SYSTEMSTATUS}" = "SUCCESS" ]?ivr1,1)
exten => username,n,System(echo \'${SIP_HEADER(TO)}\' | egrep "sip:.*66666666666@" > /dev/null)
exten => username,n,GotoIf($[ "${SYSTEMSTATUS}" = "SUCCESS" ]?ivr2,1)
exten => username,n,System(echo \'${SIP_HEADER(TO)}\' | egrep "sip:.*8888888888@" > /dev/null)
exten => username,n,GotoIf($[ "${SYSTEMSTATUS}" = "SUCCESS" ]?ivr3,1)
exten => username,n,Hangup


exten => ivr1,1, Playback(ivr-1)
exten => ivr1,2, Dial(SIP/10001)
exten => ivr1,3, Hangup()

exten => ivr2,1, Playback(ivr-2)
exten => ivr2,2, Dial(SIP/10002)
exten => ivr2,3, Hangup()

exten => ivr3,1, Playback(ivr-3)
exten => ivr3,2, Dial(SIP/10003)
exten => ivr3,3, Hangup()





[pbx-internal]

exten => _09.,1,Set(CALLERID(name)='666') ; make sure you dont use any space character
exten => _09.,2,Dial(SIP/allo.com/${EXTEN:2},120,r) ; to dialout through allo.com

; Some of the PBX internal SIP extensions
exten => 101,1, Dial(SIP/101)

exten => 201,1, Dial(SIP/201)




Click here
Image
to download and run the perl script, which gives you sorted order of proxies based on your location.

Asterisk Configuration Examples


Incoming and Outgoing Settings



here's a basic asterisk configuration to place and receive calls through allo.com network.

for the sake of simplicity, lets replace your sip.conf file by the following (pls make a backup!!):


[general]
context=default ; default context
srvlookup=no ; sip srv lookup applies to allonetwork.net, and not to specific proxy IPs
allowguest=no ; disallow unidentified sip calls to come in, useful for debugging

NOTE
dont use externip for NATings since its taken care by proxy itself.

  1. include sip_additional.conf


then lets create sip_additional.conf file, in which we'll keep the peer and registration configs:


register = username:password@67.17.211.246/username
; fastest (and closes) proxy to me is in NYC (get proxies on www.allo.com/asterisk/)

;Allo.ComTrunk Setting
[allo.com]
type=peer
username=username
fromuser=username
secret=password
context=from-allo
host=67.17.211.246
fromdomain=allo.com
insecure=very
disallow=all
canreinvite=no
dtmfmode=inband
allow=ulaw
allow=alaw


now, lets create a context for the incoming and outgoing calls of this peer, in extensions.conf



; Note: Make sure you use the '/username' at the end of register line in your sip configuration.

[from-allo]
exten => username,1,Goto(default,s,1) ; assuming you have a default context where incoming calls are routed

;or use the following dialplan to route based on the DID's

[from-allo]

exten => username,1,System(echo \'${SIP_HEADER(TO)}\' | egrep "sip:.*5555555555@" > /dev/null)
exten => username,n,GotoIf($[ "${SYSTEMSTATUS}" = "SUCCESS" ]?ivr1,1)
exten => username,n,System(echo \'${SIP_HEADER(TO)}\' | egrep "sip:.*66666666666@" > /dev/null)
exten => username,n,GotoIf($[ "${SYSTEMSTATUS}" = "SUCCESS" ]?ivr2,1)
exten => username,n,System(echo \'${SIP_HEADER(TO)}\' | egrep "sip:.*8888888888@" > /dev/null)
exten => username,n,GotoIf($[ "${SYSTEMSTATUS}" = "SUCCESS" ]?ivr3,1)
exten => username,n,Hangup


exten => ivr1,1, Playback(ivr-1)
exten => ivr1,2, Dial(SIP/10001)
exten => ivr1,3, Hangup()

exten => ivr2,1, Playback(ivr-2)
exten => ivr2,2, Dial(SIP/10002)
exten => ivr2,3, Hangup()

exten => ivr3,1, Playback(ivr-3)
exten => ivr3,2, Dial(SIP/10003)
exten => ivr3,3, Hangup()





[pbx-internal]

exten => _09.,1,Set(CALLERID(name)='666') ; make sure you dont use any space character
exten => _09.,2,Dial(SIP/allo.com/${EXTEN:2},120,r) ; to dialout through allo.com

; Some of the PBX internal SIP extensions
exten => 101,1, Dial(SIP/101)

exten => 201,1, Dial(SIP/201)




Click here
Image
to download and run the perl script, which gives you sorted order of proxies based on your location.
Created by: jac_allo, Last modification: Thu 05 of Jul, 2007 (10:19 UTC) by varadhan
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