Hello there fellow Asterisk users and developers.
From start I have to mention I am new to Asterisk, but I am trying to learn as much as possible.
What do I have
I need to deploy in our office 11 VoIP phones by Cisco - SPA502G.
We have a NAS by Synology - DS1511+ that has Asterisk installed as a package.
We have a SIP account from our provider.
Asterisk version is 1.8.13.1, Asterisk GUI-version : 2.1.0-rc1
Cisco VoIP phones updated to the latest version available.
What do I need
Setup the SIP account on Asterisk server
Setup the user extension
Make and receive calls
What I did
I have installed Asterisk package and I am able to access it
The server on which Asterisk resides has two network cards. One NIC is assigned a public IP and is connected directly to the internet, without firewall. One NIC is assigned a private IP and is connected to the local network, behind NAT and firewall.
The Cisco VoIP phones are connected to the local network and I can access their web server
What is not working
While setting up the SIP account from our provider, I get the Status “Requesting” written in red and It does not register.
Our provider stated that their platform has problems with Asterisk 1.8 and we should setup the SIP account without user name and password; only host
I created two test users on Asterisk like:
fullname=TestUser1
registersip=no
host=dynamic
callgroup=1
mailbox=203
call-limit=100
type=peer
username=203
transfer=yes
callcounter=yes
context=DLPN_DialPlan1
cid_number=203
hasvoicemail=yes
vmsecret=test
email=
threewaycalling=no
hasdirectory=no
callwaiting=no
hasmanager=no
hasagent=no
hassip=yes
hasiax=no
secret=test
nat=yes
canreinvite=no
dtmfmode=rfc2833
insecure=no
pickupgroup=1
disallow=all
allow=ulaw,gsm
macaddress=203
autoprov=yes
label=203
linenumber=1
LINEKEYS=1
I have setup the account on the Cisco SPA502G phone, but I constantly get error 404 (on the phone)
I tried various settings and options, but all lead to the dreaded 404 error.
The questions
- What might be causing this 404 error and how can I fix it?
- If the SIP account is setup on the Asterisk box but no phones are setup yet, when I call the SIP number, should I hear the line ringing on the calling side? At the moment I do not hear any rings and the call just drops, without any notice (busy, etc)
sip.conf
context=default
allowoverlap=yes
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
subscribecontext=default
allowexternaldomains=yes
allowguest=yes
allowsubscribe=yes
allowtransfer=yes
alwaysauthreject=no
autodomain=yes
bindaddr=0.0.0.0
bindport=5060
callevents=no
checkmwi=10
compactheaders=no
defaultexpiry=120
dumphistory=no
externrefresh=10
fromdomain=domain.com
g726nonstandard=no
jbenable=no
jbforce=no
jblog=no
maxcallbitrate=384
maxexpiry=3600
minexpiry=60
mohinterpret=default
notifyringing=yes
pedantic=no
progressinband=never
promiscredir=no
realm=domain.com
recordhistory=yes
registerattempts=0
registertimeout=20
relaxdtmf=no
sendrpid=no
sipdebug=yes
t1min=100
t38pt_udptl=no
tos_audio=none
tos_sip=none
tos_video=none
trustrpid=no
useragent=Voice Server
usereqphone=no
videosupport=no
dtmfmode=rfc2833
nat=no
externip=XX.XX.XX.226
localnet=10.70.1.0/255.255.0.0
allow=ulaw,alaw,gsm,ilbc,speex,g726,adpcm,lpc10,g729,g723,h263,h263p,h264
Thank you very much