[msun]

Hi Patwa,

Ok, by changing display mode for this thread to linear, I can now see the missing posts.

Re your earlier posting the basic dial plan string (8xxxxxxx<:@gw1>|xx.<:@gw0>) seems to say that any 8 digit phone number beginnning 8 should route via GW1. I think any 8 or more digits beginning 8 will go via GW1 Is it that aspect that does not work for you?

Anyway, you can usefully check out your dial plan here http://supremeit.com/voip/dialplan.php

Hope this is of some help to you.

Cheers from Mike

[trooper71]

Could someone please help me create a simple dialplan string.

I have a sipura 3102 that i will be primarily using with voxalot. All i need are the following situations to be handled by the dial plan...

• If i dial an arbitrary number of digits, it should use the PSTN line.
• If i dial # followed by an arbitrary number of digits, it should use Line 1 (which is set to voxalot).

Appreciate any help.

[hwittenb]

Originally Posted by trooper71

[*]If i dial an arbitrary number of digits, it should use the PSTN line.[*]If i dial # followed by an arbitrary number of digits, it should use Line 1 (which is set to voxalot).[/list]Appreciate any help.

Try this:
(<#:>[x*][x*].|xx.<:@gw0>)

[x*] means either a digit 0-9 or an * character.

[trooper71]

Originally Posted by hwittenb

Try this:
(<#:>[x*][x*].|xx.<:@gw0>)

[x*] means either a digit 0-9 or an * character.

Thanks a bunch hwittenb.... I was initially using something similar that i crafted but could never get PSTN service to work. Its been driving me crazy for the last couple of days.

When I lift the hook of the phone connected to the sipura box, i receive a dialtone from the ATA and not my POTS providers dialtone. When i dial a local number (e.g. 050-5142369), it does ring however i think its going through the VoIP since my cellphone never rings (i am using it as a test). No one ever picks up so i dont know where it is being routed.

The reason I think that PSTN isnt working is because the Line LED on the ATA does not come on.

The voltage read by the ATA when off hook is 65-66V and when I pick up the receiver of another phone in my home it reads 6-7V.

What should I set the Line-In-Use voltage to? The rest of the settings are currently set to default (latest firmware).

I have been reading different posts for days to no avail. I am not sure whether the problem lies in the voltage settings or the impedance settings or something else entirely. Can it have anything to do with my telecom provider's tone settings (maybe the disconnect tone or something)??

I hope someone here has the answers... oh and I live in Dubai if anyone knows the right settings.

Thanks in advance.

[hwittenb]

Originally Posted by trooper71

When I lift the hook of the phone connected to the sipura box, i receive a dialtone from the ata and not my POTS providers dialtone.

That's normal. That's the way it works. You dial the digits before the SPA takes the pstn line off hook.

When i dial a local number (e.g. 050-5142369), it does ring however i think its going through the voip since my cellphone never rings (i am using it as a test). No one ever picks up so i dont know where it is being routed.

If you are using the dial plan above, i.e. (xx.<:@gw0>), the call is not going to voip.

The reason I think that PSTN isnt working is because the Line LED on the ATA does not come on.

I don't have a SPA3102, just a SPA3000 so I don't know how the led's work.

The voltage read by the ATA when off hook is 65-66V and when I pick up the receiver of another phone in my home it reads 6-7V.

The pstn line has to be truly off hook. Just taking the handset off hook doesn't make the pstn line off hook until the SPA takes it off hook before it dials the number.

What should I set the Line-In-Use voltage to? The rest of the settings are currently set to default (latest firmware).

The default value of 30 should be fine for you if the pstn line on hook is 65v and off hook it is 7v.

Can it have anything to do with my telecom provider's tone settings (maybe the disconnect tone or something)??

If the SPA gets what it thinks is a disconnect signal, i.e. a CPC signal or disconnect signal when it takes the line off hook, that will cause trouble. After you try a call and it fails, look at the SPA's Info tab for the "Last PSTN Disconnect Reason". If it says CPC or Disconnect tone then that could be causing the call failure. There is a setting you can disable to confirm the problem solution.

[Bergling]

Originally Posted by patwa

Hello,

I'm having some trouble with my dial plan. The main problem is that nothing's going out the PSTN line. The secondary issue is that I can't work out how to add rules for diali=ling * and # codes for my PSTN provider's calling features.

Here's the requirements:
All numbers starting 8 and with 7 more digits (e.g 84459068) sent out gw1
All other numbers regardless of length, including those starting with 8 (but not 8 with seven digits) sent out PSTN
Numbers with the form *xx# sent out PSTN (UK BT calling features)
Numbers with the format #xx# sent out PSTN (UK BT calling features)
Numbers with the format *#xx# sent out PSTN (UK BT calling features)
If this can be implemented to begin with, there is no need for dedicated gateway accessors, e.g #9 for PSTN or #1 for gw1.

Therefore, examples of numbers that should be able to be dialled, and sent through unmodified to PSTN are:
1471, 999, 123, 01224783228,*61#, #21# *#21#

The basic plan therefore, ignoring the * and # codes for the moment should work:
(8xxxxxxx<:@gw1>|xx.<:@gw0>)

But this does not work. Further, I have read this thread for how to sent the * and # as digits to PSTN, but have either missed or misunderstood the details if they were here.

For info, I have set all the calling features on the Line1 tab to no, e.g call forward, block, accept, DND, conference, etc.

Any help appreciated.

Thanks.

H.

I would also appreciate help with figuring out how to use the '#' sign for dialing my voicemail. My VOIP-provider has *133# for accessing voicemail, but I still haven't figured out how to fix that in my dial plan.

[hwittenb]

Originally Posted by Bergling

I would also appreciate help with figuring out how to use the '#' sign for dialing my voicemail. My VOIP-provider has *133# for accessing voicemail, but I still haven't figured out how to fix that in my dial plan.

Did you try an element like |*133#| in your dial plan or |*133#S0|. If so, what happened?

Does your provider require that you send the # sign? If you are not supposed to send the # then you would leave it out of the dial plan. Unless the # is in the dial plan, it is sort of a special character that tells the SPA to send what it has immediately.

[Bergling]

Actually I tried with the |*133#S0| and it worked.

Thanks.

[voip_addict]

Not sure how many times you can rewrite to an ATA before eventually killing it. Over the course of the last 7 months I've been trying differen things with my dial plans on a few of my devices I'm afriad it only a matter of time before one of them has had enough abuse and just quits. So, before overwriting the dial plan yet another time I wanted to ask.

At the end of my dial plan I currently have
[*x][*x].<:@sipbroker.com>

The problem with this is it doesn't allow calls to numbers like
*393*1800xxxxxxx but will allow other numbers such as *39355555

The reason this comes up is because I usally use sipphone for 800 numbers but the call quality is sometimes better using fwd for toll free when calling from my area so I like to have that option.

So to overcome this problem and be able to dial numbers like *393*xxxxxxxxxx would I be better off talking [*x][*x].<:@sipbroker.com> and adding *xxx.*x.<:@sipbroker.com>| before it?

Or should I replace
[*x][*x].<:@sipbroker.com>

with
[*x].[*x].<:@sipbroker.com> (period between brackets)

or
[*x]xxx.[*x].<:@sipbroker.com> ?

[hwittenb]

Originally Posted by voip_addict

[*x][*x].<:@sipbroker.com>
The problem with this is it doesn't allow calls to numbers like
*393*1800xxxxxxx

I believe your original entry should work unless it is ambiguous with some other entry in your dial plan. I tested it with my SPA3000, as the only entry in the dial plan, and it sent the call to SipBroker and the call completed. I captured it with ethereal and it did send it to SipBroker:
21 0.131508 192.168.1.100 64.34.162.221 SIP/SDP Request: INVITE sip:*393*18005069511@sipbroker.com, with session description

edit: the [*x][*x]. says you can have either * or a digit for any character and the . (dot) says you can repeat the sequence any number of times.

Maybe I'm missing something.

As to re-writing the configuration on the ata, I've done that thousands of times.