FXO establishes connection but gets busy tone? :?

[WolfricElkrose]

1. Brekeke Product Name and version:
PBX version 2.1.6.6
2. Java version:

3. OS type and the version:
Windows Server 2003

4. UA (phone), gateway or other hardware/software involved:
Xlite/Cisco 7960series/ SpectraLink Ptx150's/ Grandstream GXW4108
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :
Pattern 2 with no FXS.

6. Your problem:

Seems all is fine, I access the FXO with everything fine:
Dial Plan:
T0=sip:9(.*)@ Deploy To=sip:$1@192.168.8.12 <--Grandstream

Basically when the call goes through in the Call Log on the SIPSide shows that it went through successfully.

43 sip:1007@192.168.8.5:5060 sip:5233469@192.168.8.1 Success -1

It dials, sometimes a brief ring then busy tone, or hangs up.

Not sure what I'm missing on the FXO side really, but I'm positive the PBX side is dandy

Not sure if there is anything in specific that I'm missing on the Grandstream gxw4108.

[taitan]

HI

> Dial Plan:
> T0=sip:9(.*)@ Deploy To=sip:$1@192.168.8.12 <--Grandstream

Is it DialPlan? or Is it ARS??
If it is DialPlan, "$1" should be "%1"..

> Not sure what I'm missing on the FXO side really,

Which audio codec are you using at the FXO and PBX?
Are you using the same codec there?

[WolfricElkrose]

Hey taitan, thanks for the quick response!!

>Is it DialPlan? or Is it ARS??
ARS I have:
To:sip:9(.+)@ To:sip$@192.168.8.12 <--GrandstreamFXO

DialPlan changed from:
$request=^invite
To=sip:9(.+)@ To=sip:$1@192.168.8.12

Changed as per your suggestion to:
$request=^invite
To=sip:9(.+)@ To=sip:%1@192.168.8.12

Not sure what the difference is between the $ or % in this case, if anyone can explain??

>> Not sure what I'm missing on the FXO side really,
>Which audio codec are you using at the FXO and PBX?
>Are you using the same codec there?

On the FXO side: only using Profile 1 with PCMU and PCMA selected. As for the PBX side, I'm not sure what your referring to on the PBX side if you can clarify maybe?

The only place I find Settings for Codexs on the PBX side is in Options, but I'm not sure what to put in there.

Thanks for your help thus far Now I'm just confused on the audio codex part

Wolf

[taitan]

> $request=^invite

Ummm... It should be "$request=^INVITE"..


> On the FXO side: only using Profile 1 with PCMU and PCMA
> selected. As for the PBX side, I'm not sure what your referring
> to on the PBX side if you can clarify maybe?

Brekeke PBX supports PCMU and PCMA. So you don't have to worry about codec.

[WolfricElkrose]

my fault I guess. It is INVITE, I was too lazy to hold the shift key when I retyped it in my last post.

So k, so the the ARS and the Dial Plan are fine, the PCMU and PCMA settings are fine for the FXO side.

Just so to clarify what I have set that I'm not sure about:
Using 1 Stage Dialing, in the North America.

STARTING with the FXO side

The hardware: Grandstream GXW4108
(8 FXO Ports, but only using 1 for now, trying to get it working first before getting to complex)
Settings:
FXO Lines-Wait for Dial-Tone Yes, Stage Method 1, Delay before Dial PSTN 100ms, T.38 Setting: ch1-8:mode=2,rate=9600,ecm=1; (mode = Passthrough)
Channels - Prefix 9, SIP is handled on PBX side
PSTN Outgoing - {x+}
Profile1 - Sip Server and Proxy points to PBX, Sip Registration No, DNS SRV No, Early Dial No
Advanced Settings, Allow DHCP Option 66 to override server No, Use random port no, Voice Frames per TX = 2
(rest are at defaults)

The PBX Side:
ARS set up properly. RTP Relay default
Options-Settings(all on defaults) RTP relay off, Session Keep Alive 600

The Sip Server Side:
Dial Plan set up properly.
Configurations
System- defaults
SIP - REGISTER ON, INVITE ON, Auth-To:no, Auth-From:no, Thru Reg on
RTP - RTP Relay on
Database - default
Advanced - nothing


Hopes this helps Anything not defined is at their default settings. I'm sure I'm overlooking something small, but just can't seem to spot it.

Thanks for all the help I can get,
Wolf

[hope]

From the post below
http://www.brekeke-sip.com/bbs/viewtopic.php?t=6243&highlight=grandstream+gxw4108
Use prefix# (default is 99) + ch# (could be anything from 1 to 8) + dialing# will result in this call forwarded to FXO port (ch#) immediately.

And i don't think you need both dialplan and ARS for dialing to GW

[WolfricElkrose]

[lakeview]

>>It dials, sometimes a brief ring then busy tone, or hangs up.

Do you have the same problem still?
Can you make a call between X-Lite and SpectraLink Ptx150?

[WolfricElkrose]

[lakeview]

Can you make a call from the outside to a SIP client via FXO gateway?

[WolfricElkrose]

[lakeview]

Wolf,

Yeah,, I agree with you.

>>It dials, sometimes a brief ring then busy tone, or hangs up.

Let me know more details..
If you dial the outside (your cellphone for example) from the SIP-side, you can hear the ring tone on the caller. Right?
At that time, does the outside's phone ring too?

[WolfricElkrose]

[lakeview]

Hi Wolf,

If you set "2 stages-Dialing" at the Grandstream, can you make a call to an outside?

Also, check your Grandstream's firmware version. It should be the latest.

[WolfricElkrose]

[lakeview]

With "2 stages-Dialing" mode, can you hear any tone after you connect to the Grandstream from a SIP client?
At that time, can you dial an outside phone number?

[tagaamo]

Hi Wolf

hope this can help.....

using GXW-4108 PLEASE CREATE ASR remove your dial plan
Patterns - OUT only disable Patterns - IN

Matching patterns
TO: sip:9(.*)@

Deploy patterns
TO: sip:$1@192.168.0.2

click SAVE and UPDATE

where 192.168.0.2 is the IP adress of GXW-4108

on GXW-4108 go to FXO Lines menu

set 1 stage dialing
set Unconditional Call Forward to VOIP: ch1-8:100
where 100 is your auto attendant

save your setting and restart GXW

now to call outside line ex. Tel No. 1234567 dial 91234567

It is recomended that you use 1 stage dialing cause when incoming call from PSTN phone must go directly to auto attendant, they must not hear dialtone.

[tagaamo]