I am very new to asterisk, I just set a pbx up in my basement. I have got several extensions working with ip phones, and soft phones. I recently got a stanaphone free DID so that I can call my pbx from a PSTN line. I can not figure out how to set up my pbx to recieve the DID call and ring my extensions.
As of now I am getting this message on my asterisk console with verbosity 5.
– parse_srv: SRV mapped to host sip.stanaphone.com, port 5060
== Parsing ‘/etc/asterisk/sip_notify.conf’: Found
– parse_srv: SRV mapped to host sip.stanaphone.com, port 5060
– Got SIP response 503 “Service Unavailable: cli is wrong, check your configuration, switch to 08 number” back from 204.147.183.18
here is a snip from my sip.conf
//—begin sip.conf
[general]
context=default
srvlookup=yes
context=sip ; Default context for incoming calls
register => my-10-digit-DID:my-stanaphone-pass@sip.stanaphone.com/101
; This section is because i’m behind nat
externip=tamnet.linuxhome.org;Outside address
localnet=192.168.1.55 ;Inside address
localmask=255.255.255.0 ;Inside subnet
[stanaphone]
type=friend
username=my-10-digit-DID
secret=my-stanaphone-pass
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
;gxp-2000 line 4
[2000]
accountcode=2000
type=friend
secret=blah
qualify=no
nat=no
host=dynamic
canreinvite=no
context=internal
//—end sip.conf
I have an an entry for my gxp-2000 in extensions.conf… but i dont have anything for stanaphone. Basically im a total newb to *, and lack the problem solving skills to figure this out. I really appreciate any help.
-Eric