FXO establishes connection but gets busy tone? :?

WolfricElkrose · Post by **WolfricElkrose** » Tue Oct 07, 2008 11:54 am

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/ ... terns.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 · Post by **taitan** » Tue Oct 07, 2008 8:57 pm

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 · Post by **WolfricElkrose** » Wed Oct 08, 2008 5:25 am

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 · Post by **taitan** » Thu Oct 09, 2008 12:27 am

> $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 · Post by **WolfricElkrose** » Thu Oct 09, 2008 5:35 am

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 · Post by **hope** » Fri Oct 10, 2008 9:48 am

From the post below
http://www.brekeke-sip.com/bbs/viewtopi ... am+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 · Post by **WolfricElkrose** » Tue Oct 14, 2008 6:30 am

hope wrote:From the post below
http://www.brekeke-sip.com/bbs/viewtopi ... am+gxw4108
Use prefix# (default is 99) + ch# (could be anything from 1 to + 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

Thanks for the quick reply hope, I did see that post, but niggy never did say how he resolved the problem. The problem isn't the prefix etc, there's more to it than that, and niggy figured it out. I'm in the same process of turning certain settings on/off, etc, but nothing seems to kick in properly.

I just wished when people have thier problems solved

that they post what they did to solve it, or if someone's solution actually helped or not

Wolf

lakeview · Post by **lakeview** » Tue Oct 14, 2008 10:35 am

>>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 · Post by **WolfricElkrose** » Tue Oct 14, 2008 12:16 pm

lakeview wrote:>>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?

yeah that's the thing, I can call everything but through the FXO.

I have an X-lite setup, the SpecraLink Ptx150 and a Cisco IP Phone 7960 series, and I can call each one and vise versa. As for hitting outside, I'm kinda stumped.

Now if I use a normal phone directly into the line, no FXO, I have to dial 9+number to reach the outside line. That works fine.

Tried 9+1+9number still doesn't seem to want to go through, so I'm assuming its on the FXO side. I am positive I'm missing something really small, besides my sanity

Wolf

lakeview · Post by **lakeview** » Tue Oct 14, 2008 3:45 pm

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

WolfricElkrose · Post by **WolfricElkrose** » Wed Oct 15, 2008 5:29 am

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

Um no, I don't have it set up to do so, I'm more concentrating on getting it to dial out first, then figure out the dial in. Doing both would be harder to trouble shoot. Same reason why I'm using the one stage dialing.

Start small to get it working properly first than move on to more challenging tasks. Easier to trouble shoot issues, when you know the basics are working.

I am new to the whole FXO side of things, I understand the whole SIP/PBX side of things at an advance level, no way will I boast expert level

not till I know exactly what's going on.

I'm using wireshark as well to watch what's going on, but it doesn't seem to show any errors as it hits the FXO side either, which is why I believe it's how I have the FXO side set up, the manual for the Grandstream just tells you the bare minimum. Anyone who dealt with it, probably may agree. The manual does explain if you are using SIP but not on the FXO side. (ie anything similar to Brekeke or even Asterisk)

Hope this clears up what I'm asking?

Wolf

lakeview · Post by **lakeview** » Wed Oct 15, 2008 10:53 am

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 · Post by **WolfricElkrose** » Fri Oct 17, 2008 5:33 am

lakeview wrote: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?

Nope, tried an outside phone, still doesn't go through, seems to say 'success' on the Sip log side of things, so the dialplan seems to be fine. And no, it doesn't ring on the outside caller phone, which makes my issue more depressing

heh.

So my question being, if internal is working, therefore it has to be the Grandstream end. Bah, even browsing the Grandstream forums and still no one to answer on my issue. I'm positive it is a setting or something I didn't set up on the Grandstream side.

Wolf

lakeview · Post by **lakeview** » Fri Oct 17, 2008 11:04 am

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 · Post by **WolfricElkrose** » Fri Oct 17, 2008 12:42 pm

lakeview wrote: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.

Hmmm, switched to 2 stages, still no go.

The firmware is up to date, I swear there's something out that I'm missing, and it's a really small thing too

Bet when I figure it out, I'll be ready for the crazy room....

Wolf

lakeview · Post by **lakeview** » Fri Oct 17, 2008 1:41 pm

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 · Post by **tagaamo** » Fri Oct 24, 2008 7:06 pm

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 · Post by **tagaamo** » Fri Oct 24, 2008 7:44 pm

WolfricElkrose wrote:
lakeview wrote: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.
Hmmm, switched to 2 stages, still no go.

The firmware is up to date, I swear there's something out that I'm missing, and it's a really small thing too

Bet when I figure it out, I'll be ready for the crazy room....

Wolf

If you want to set to 2 stage dialing you must first create user on sip server and set SIP Registration: YES

it must be regestered to the sip server then dial that No. and then you get dialtone

ex. create user on SIP server ex. 100 and set password too

then go to GXW on Channel menu
ex. Channel 1
SIP User ID: 100
Authenticate ID: 100
Authen Password: your password here
Profile ID: profile 1

then go to Profile 1 menu

SIP Server: your SIP server
SIP Registration: YES

restart GXW

be sure that PSTN line is connected to channel 1 of GXW channel

now try to dial 100 from your sipphone you should hear dialtone from PSTN line then you can dial 7 digit outside No.

FXO establishes connection but gets busy tone? :?

FXO establishes connection but gets busy tone? :?

Ummm yeah