Hello All,
I have managed to get my A580ip connected to asterisk, however the calls drop within 3 to 10 seconds.
This doesn’t happen from a softphone like x-lite so it must be some configuration issue with the A580ip.
I am running asterisk on a ubuntu server hosted on a virtual machine on the internet. Therefore my clients
are all behind a NAT router.
I have the following in my sip.conf:
[ul]
[general]
context=default
subscribemwi=yes
allowoverlap=no
tos=0x18
bindport=5060
bindaddr=0.0.0.0
externip=98.156.123.127
nat=yes
externhost=tigger.domain.co.uk
externrefresh=120
localnet=98.156.123.127/255.255.255.0
srvlookup=yes
disallow=all
allow=alaw
allow=ulaw
allow=g726
allow=ilbc
allow=g729
allow=gsm
[1000]
type=peer
host=dynamic
context=internal
secret=1000
callerid=FBI
dtmfmode=rfc2833
qualify=yes
nat=yes
mailbox=1000
[1001]
type=peer
host=dynamic
context=internal
secret=1001
callerid=FBI2
dtmfmode=rfc2833
nat=yes
mailbox=1001
[/ul]
My externsions.conf:
[ul]
[general]
static=yes
writeprotect=no
[internal]
exten => 55,1,Playback(demo-echotest)
exten => 55,2,Echo
exten => 55,3,Playback(demo-echodone)
exten => 1000,1,Dial(SIP/1000,20)
exten => 1001,1,Dial(SIP/1001,60)
[/ul]
I have a feeling it could be an issue with the localnet setting.
Any help on getting this working, or suggestions on where to look would be very much appreciated.
Thanks,
James
Hi James
Set canreinvite=no
also just use externip
what base are you using as teh a580ip is just the handset, I guess its teh n300
what are your settings in that? we have many Gigasets out there all working flawlessly
Hi Ian,
Many thanks for the helpful response.
Actually the A580IP is a base-station and handset combo (comes with a A58H handset). The base-station is very similar to the N300, but a bit older. I couldn’t work out the differences to be honest, but would like to have a full voicemail solution so I’m setting up asterisk. I might upgrade to the N300A at a later date.
I just adjusted to:
[ul][general]
context=default
subscribemwi=yes
allowoverlap=no
tos=0x18
canreinvite=no
bindport=5060
bindaddr=0.0.0.0
externip=98.156.123.127
srvlookup=yes
disallow=all
allow=alaw
allow=ulaw
allow=g726
allow=ilbc
allow=g729
allow=gsm[/ul]
It’s still the same problem unfortunately. The handset works for a few seconds and then It beeps displaying “VoIP status : 801” on the screen.
So my gigaset settings: …
For the account:
[ul]
== Personal Provider Data ==
Authentication Name: 1000
Authentication Password: 1000
Username: 1000
Display name: zog
== Provider Information ==
Domain: asterisk
Proxy server address:
Proxy server port: 5060
Registrar server: 98.156.123.127
Registrar server port: 5060
STUN DISABLED – Can I set up a STUN server to help??
Outbound proxy mode: Auto
Outbound proxy: 98.156.123.127
Outbound proxy port: 5060
[/ul]
(Note about my previous post, there was a mis-match of IP’s for externip and internalnet, this was because I was obscuring my real IP and forgot to change 1, you can assume I really use the correct ones).
Advanced settings:
I’m going to play around a bit more, hope you can spot something in my configuration.
Thanks,
James
