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Grandstream 386 ATA

hobjohns
Dabbler
Posts: 22
Registered: 30-07-2007

Grandstream 386 ATA

I'm having no success getting my Grandstream 386 to connect to PlusTalk. I've seen the thread on the Grandstream phone, and tried to use the same principles to complete the settings for my 386, but still can't connect. I have my 386 working perfectly with Sipgate.

I have tried FXS port settings for PlusTalk as follows

SIP Server: myusername.plus.com
Outbound Proxy: sip.plus.net
SIP User ID: My PlusTalk SIP ID
Authenticate ID: My PlusTalk SIP ID
Authenticate Password: My PlusNet logon password
Name: Ryder Cowan
Register expiration: 60
local SIP port: 5060
local RTP port: 5004
Enable call features: No
Send DTMF: via SIP INFO
DTMF payload type: 101

Under General Miscellaneous Settings (applicable to all FXS ports) I've also set Preferred Vocorder to G729 and for Sipgate I have had NAT Traversal set to Yes, with STUN server as stun.sipgate.net:10000. Can I leave the STUN server setting as it was, or will PlusTalk require something different? (If different, that of course presents problems if I want to retain my Sipgate service on one FXS port of the 386, and have PlusTalk on the other.)

I'm obviously missing something - can anyone suggest what else I need to do to connect?

Ryder Cowan
9 REPLIES
shamus
Grafter
Posts: 114
Registered: 13-08-2007

Grandstream 386 ATA

I dont have a Grandstream but I did spend weeks trying to get a Sipura 3000 working...

I'm pretty sure that you need to set:

SIP User ID: plustalk
Authenticate ID: plustalk

This thread may help (not sure if this is the one you read or not) http://portal.plus.net/central/forums/viewtopic.php?t=32175

HTH

Shaun
hobjohns
Dabbler
Posts: 22
Registered: 30-07-2007

Grandstream 386 ATA

Very many thanks - that did the trick.

I've found that I do need to keep the NAT traversal setting to 'Yes, and specify a STUN server - PlusTalk appears to work with the Sipgate STUN server address. Do you know if PlusTalk publish a STUN server address?
N/A

AT320

Sorry I won't be able to help you but I am hoping you may be able to assist me?
I have an AT320 VoIP phone plugged into a Netgear DG834v2 ADSL firewall router.
I registered with Sipgate but get "Log On Failed" (if I'm lucky), every time.
Netgear have not answered my questions & even the Sipgate help does not cover this particular phone or router.
The AT320 phone has Plusnets DHCP'd IP which pings OK.
If I use PlusNets DHCP to allocate the phone IP, do I use PlusNets dns IP's?

Sipgate help advised "forward the needed ports" (I do not understand)
Port: 5060/UDP (SIP signalling)
Port: 5004/UDP (RTP, language)
Port: 8000-8012 (Voice)
Port: 10000 UDP (STUN)
Port: 3478 UDP & TCP "
The firewall router is set to:
OUTBOUND SERVICES are "ALLOW always"
INBOUND SERVICES are "BLOCK always"
does "forward" relate to allowing specific INBOUND SERVICES?

Although I've now got a Plus Talk ID I want to get the Sipgate account working 1st, are you connecting to Sipgate & Plustalk on the same handset?
hobjohns
Dabbler
Posts: 22
Registered: 30-07-2007

Grandstream 386 ATA

Have you tried allocating a fixed IP address to your ATA and placing that address in your router's DMZ? That ought to deal with any port-forwarding requirements.
N/A

AT320

Thanks for responding promptly hobjohns.
The router had uPnP enabled already so I allocated a staic IP to the VoIP phone & added it as a DMZ server to the router, I also added a static route but still get "Log On Failed".
Must be my settings/addresses, I'd like someone with an AT320 connected to Sipgate to give an example list of the phone settings i.e.

NETWORK SETTINGS
iptype > static
ppp id > (blank)
ppp pin > (blank)
local ip > 192.168.*.*
subnet mask > 255.255.255.0
router ip > 192.168.*.*
dns > (is this the ip of my provider's primary dns server?)
dns2 > (is this the ip of my provider's secondary server?)
mac > (VoIP phones mac address)

AUDIO SETTINGS
codec1 > gsm
codec2 > null
codec3 > null
codec4 > null
codec5 > null
codec6 > null
vad > (unchecked)
agc > (unchecked)
aec > (checked)
audio frames > 2
jitter size > 0
g.723.1 high rate > (checked)

PHONE SETTINGS
use dialplan > disable
dial number > (blank)
ddd code > 10
idd code > 86
idd prefix > 00
ddd prefix > 0
inner line > disable
inner line prefix > 0
call waiting > (unchecked)
forward number > (blank)
fwd poweroff > (unchecked)
fwd noanswer > (unchecked)
fwd always > (unchecked)
fwd busy > (unchecked)
answer > 30
use digitmap > (unchecked)
handset in(0-15) > 7
hanset out(0-31) > 20
ring type > dtmf
speaker out(0-31) > 20
speaker in(0-15) > 4

SIP PROTOCOL
use service > (checked)
register ttl > 3600
service type > sipphone
sip proxy > sipgate.co.uk
domain/realm > sipgate.co.uk
nat traversal > stun
nat addr > stun.sipgate.net:10000
nat ttl > 3600
phone number > 01*********
account > 5898***
pin > V*******
register port > 5060
rtp port > 5004
rtp tos > 0
call type > normal
dtmf > sip info
dtmf payload > 101
super password > ********
debug > diable

OTHER SETTINGS
password > ****
upgrade type > disable
upgrade addr > (blank)
sntp ip > (even when set blank, this field obtains an IP after rebooting the phone but I don't recognise it?)
use daylight > (checked)
timezone > GMT

If someone with an AT320 reads this, maybe they could point out what I've set incorrecly? Much appreciated!
hobjohns
Dabbler
Posts: 22
Registered: 30-07-2007

Grandstream 386 ATA

Have you tried amending the codec settings? The Sipgate website suggests:-

PCMA
PCMU
G729
G723
G726

for codec options 1 to 5.
N/A

Grandstream 386 ATA

Thanks hobjohns

I've used the codec settings from a great link to a screen shot at
http://www.aredfox.com/esetsip8.htm.
Have now gone back to a DHCP'd IP with no DMZ or static route, it was just my bad settings.
Am still interested to know if you are connecting to Sipgate & Plustalk on the same handset?
hobjohns
Dabbler
Posts: 22
Registered: 30-07-2007

Grandstream 386 ATA

Same ATA, with two FXS ports, one to PlusTalk, one to Sipgate.
N/A

Grandstream 386 ATA

I had real problems getting any VOIP phone working, In the end we had to change port 5060 to 5080, it looks like +net reserve 5060 for themselves (although they have denied this) so if you are using a provider other then +net it is worth trying. Mine now works fine after I did this, but it did take a couple a weeks to figure out (it is a business line as well!!!!)