HT-503 not outbound calls

[aesteve]

Hello,

we´re configuring a Grandstream HT-503 and we´are having a problem with the outbound calls. This is the log. A 3CX locution says that the call is forbidden by the admnistrator. If you put and SPA3102 in the same conexion it works perfectly.

We have the HT-503 at wan port with an static IP. The led LINE is off.

All firewall tests ok

18:14:41.734 [CM503008]: Call(384): Call is terminated
18:14:36.796 [CM503007]: Call(384): Device joined: sip:[email protected]:40600;rinstance=7147e44283970721
18:14:36.796 [CM503007]: Call(384): Device joined: sip:[email protected]:3862;rinstance=08aaacc3a31bcdd9
18:14:36.796 [CM503002]: Call(384): Alerting sip:[email protected]:40600;rinstance=7147e44283970721
18:14:36.593 [CM503025]: Call(384): Calling Unknown:Ext.EndCall@[Dev:sip:[email protected]:40600;rinstance=7147e44283970721]
18:14:36.593 [CM503016]: Call(384): Attempt to reach <sip:[email protected]:5060> failed. Reason: Forbidden
18:14:36.593 [CM503003]: Call(384): Call to sip:[email protected]:5062 has failed; Cause: 403 Forbidden; from IP:192.168.1.123:5062
18:14:36.578 [CM505002]: Gateway:[Linea Analogica] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:]
18:14:36.578 [CM503002]: Call(384): Alerting sip:192.168.1.123:5062
18:14:36.421 [CM503025]: Call(384): Calling PSTNline:961120272@(Ln.10003@Linea Analogica)@[Dev:sip:192.168.1.123:5062]
18:14:36.375 [CM503004]: Call(384): Route 1: PSTNline:961120272@(Ln.10003@Linea Analogica)@[Dev:sip:192.168.1.123:5062]
18:14:36.375 [CM503010]: Making route(s) to <sip:[email protected]:5060>
18:14:36.375 [CM505001]: Ext.190: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 4.0.9530.0] PBX contact: [sip:[email protected]:5060]
18:14:36.359 [CM503001]: Call(384): Incoming call from Ext.190 to <sip:[email protected]:5060>

 

[mhanson]

What does your outbound dial plan look like?

 

[leejor]

Double check, triple check your password. I've had a number of problems relating to passwords being incorrect when using Grandstreams. I had to put it in five times, on one unit, to get it to work. The problem with the devices I've use is that you can't see the password as you are entering it so you can be unaware of a typo. Also...just in case, make sure you have downloaded and installed the latest firmware for the ATA, occasionally that can be an issue.

 

[igor.snezhko]

Set port number to 5062 in the 3cx PSTN Gateway settings.

 

[aesteve]

Hello and thanks for the reply :

The password is a test password 1111 , I was read somethings about Grandstream and the passwords problem. I ReWRITE the passwords a lot of times. In the status FXO and FXS are registered. The firmware is the lastest.

The sip port of the gateway in 3CX is 5062

But can´t call outbound.

More ideas ?

 

[igor.snezhko]

Seems that you calling not existing (allowed) number.

 

[aesteve]

Don´t work with any number, Normal, Mobile, etc.....


Mhanson Say :

What does your outbound dial plan look like?

Waht is this ?

 

[igor.snezhko]

Are you sure you can call from this line using analog phone? Check this first.

 

[aesteve]

Yes, it is tested and works fine with an analog telephone in this line. With HT-503 i can receive calls and 3CX works.

The unic problem is the outbounds calls.

Regards.

 

[FBW]

Today I tried the same thing with the exact same result. Anybody any idea how to solve this?

 

[leejor]

I've used a Grandstream Gateway before, but not the HT-503 (unfortunally) the Cause 403 massage is.."The server understood the request, but is refusing to fulfill it. Authorization will not help, and the request SHOULD NOT be repeated."

I would double check all of the gateway settings, obviously, it is unable to send out any call on the line. I have seen things like that happen when the phone line is seen as being busy, either because it was, or the linecord was open, or incorrect settings.

You have it set up for one-stage dialling? You've turned off any security/PIN authorization? You're not forcing the use of a Codec that 3CX doesn't support?

You may need to do a factory reset and start again from page one.

 

[willow]

Check the stage dialing on the fxo port. If it is set to 2 change it to 1.

 

[FBW]

Hi Thanks for the tip. That got me 1 step further already

However, now I get the following:
12:41:22.150 [CM503020]: Normal call termination. Reason: Server Failure
12:41:22.149 [CM503016]: Call(100): Attempt to reach <sip::<numberdialed>@<PBXIP>:5060> failed. Reason: Server Failure
12:41:22.147 [CM503003]: Call(100): Call to sip:<numberdialed>@<PBXIP>:5062 has failed; Cause: 503 Service Unavailable; from IP:10.32.1.214:5062

Is this related to the *00 dial code in the HT503?
Meanwhile I should double check that line. We haven't tried it yet with a classical phone.

Thanks again for the help.

 

[willow]

Default the box, set it back up on the HT503. then remove the pstn provider in 3cx and recreate it with the HT-488 template. I have installed a bunch of these as a cheap alternative for analog backups and they seem to work pretty good doing it this way.

 

[FBW]

Hi,

Maybe it's just the line, who knows...

I actually used the HT488 template to set up the HT503 in 3CX and it registers fine.
I only changed the following on the HT503:
- Time Zone
- WAN side HTTP/Telnet access to Yes to be able to connect to it afterwards. Or do you connect it with the LAN?
- No FXS config
- FXO: filled out the 3CX IP 3 times in the first lines and left UDP as SIp transport. Filled out the 3CX extension data and named it (unimportant).
Changed preferred DTMF method to:
P1: RFC2833
P2 In-audio
P3: SIP INFO

Codec choice 1 changed from PCMU to PCMA (is standard for Europe).
AC Termination Model to Country-base Belgium
Stage Method set to 1.

Those are the only changes made.
The only thing I'm not sure about yet is wether I should change the Device Mode from NAT Router to Bridge.
But I suppose that won't make a lot of difference here.

Again thanks for all your feedback. I think I'm nearly there :mrgreen:

 

[leejor]

The only thing I'm not sure about yet is wether I should change the Device Mode from NAT Router to Bridge.
But I suppose that won't make a lot of difference here.

If you only use the WAN port, then it won't make any difference. Setting it to Bridge (no DHCP) disables the internal router and allows you to use the LAN port, as a "pass through" with any other device (a PC?) on your network. It comes in handy if you are running short of ports on your router/bridge.

 

[FBW]

Thanks for all your help. I'll keep you informed if I get it up and running.

 

[willow]

Any luck?

 

[varietel]

It looks like your HT-503 phone is not passing all the digits. Your log is showing 961120272 a 9 digit number instead of normal 10 digits. I do not think this can cause Forbidden error but I am just curious why you are you sending a 9 digit number. I am assuming your Grandstream dial plan is not configured right. Change your dial plan to { x+|*x+|*x+*x } and see if it passes all 10 digits. May be this can help with your call going through as well. I am assuming you have set the strip digit field in the 3cx outbound rules to 0 and that is not the reason for the 9 digit number in the log. Otherwise your other pohone would not have worked either.

Take care,

Roger Partovi
Varietel Communications
Certified 3CX partner
801-889-1414
[email protected]

 

[FBW]

Hi guys,

I got it to dial out successfully.
The only change it needed, except for the 1-stage dialing, was NOT to wait for a dial tone. And now I can successfully dial out.
Thanks again for all your help. Hope this feedback will in turn help somebody else.