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Sample Asterisk config for StanaPhone

Created by: papafox,Last modification on Thu 05 of Apr, 2007 [11:04 UTC] by tommybui

Sample Asterisk config for StanaPhone


This works for me. All calls to my StanaPhone number ring on my extension 101, and after 20 seconds goes to my * voicemail. To use the StanaPhone network I dial 9 and the number.

sip.conf

 [general]
 port = 5060
 bindaddr = 0.0.0.0
 allow=ulaw

 ; This section is because i'm behind nat
 externip = 206.45.88.212 ;Outside address
 localnet = 192.168.2.10 ;Inside address
 localmask = 255.255.255.0 ;Inside subnet

 context = sip ; Default context for incoming calls
 register => stanaphonenumber:stanaphonepassword@sip.stanaphone.com/101

 [stanaphone]
 type = friend
 username = Stanaphonenumber
 secret = Stanaphonepassword
 host = sip.stanaphone.com
 context = sip
 nat = yes
 canreinvite=no ; for NAT, but it will eat up TWICE the bandwidth because everything will go through *
 insecure=very ; this will prevent * from sending a SIP 407 error back when a call comes in


 [phone1] ;x-lite
 type=friend
 username=phone1
 secret=12345678
 host=dynamic
 defaultip=192.168.2.199
 dtmfmode=rfc2833
 context=intern
 callerid="phone1" <101>
 mailbox=101



extensions.conf

 [general]
 static=yes
 writeprotect=no

 [globals]
 STANAPHONEUSERID=stanaphonenumber

 [intern]
 include => stana-out
 include => sip

 [stana-out]
 ; STANAPHONE - Dial 9 & number to use stanaphone
 exten => _9.,1,SetCallerID(${STANAPHONEUSERID})
 exten => _9.,2,Dial(SIP/${EXTEN:1}@sip.stanaphone.com,60,tr)

 [sip]
 ; phone1 - x-lite
 exten => 101,1,Dial(SIP/phone1,20,tr)
 exten => 101,2,Voicemail(u101)
 exten => 101,3,Hangup
 exten => 101,102,VoiceMail(b101)
 exten => 101,103,Hangup 


For me this above did not work, I was getting "Failed to authenticate on INVITE to" or similiar messages,
I have found working config on

http://forum.stanaphone.com/viewtopic.php?t=993&sid=000fc84c46fabe94bbd324e95963c624
http://forum.stanaphone.com/viewtopic.php?t=1327
http://forum.stanaphone.com/viewtopic.php?t=1153

Using AMP to configure Stanaphone

General SIP configuration


Comments

Comments Filter
222

333stanaphone firewall

by kFuQ, Wednesday 05 of April, 2006 [20:29:23 UTC]
i had to open tcp and udp ports 5060-5069 for stanaphone to register with asterisk

snip from sip.conf

;stanaphone
register=> 081xxxxx:xxxxxx@sip.stanaphone.com/081xxxxx

Stanaphone
username=081xxxxx
context=sip:incoming
type=peer
secret=xxxxxx
qualify=yes
nat=1
insecure=very
host=sip.stanaphone.com
dtmfmode=rfc2833
canreinvite=no


222

333Re: is this working

by , Tuesday 21 of December, 2004 [03:35:30 UTC]
Yes, It works.
Did not enter all the reqired information(:lol:)
222

333is this working

by , Monday 20 of December, 2004 [03:03:05 UTC]
is this working?
I entered my number and password following the above step. This does not work(:question:)