[uo_mba]

I've been seeking a good source for useful and interesting dial plans (other than those submitting on Voxilla.com my Phoneboy and a few other advanced users). Thus far, I haven't found such a source. (And neither has BroadVoice... when I asked them about a source, they suggested that I forward it to them so they could use it... LOL). It *may* be out there, but I just haven't since it as of yet.

Therefore, I would like to suggest the following:

1. Create a fun little Dial Plan Wizard along the same lines as the SPA wizards that Phoneboy created.

2. If a wizard isn't possible, create a section on the forum where people can share their sample dial plans, and explain how the dial plans works on their specific gateway.

3. Create an easy-to-understand summary of dial plan syntax.

If I were a programmer, I would offer to create the wizard, but, alas, I am but a simpleton (i.e. a marketing guy). I would also offer to create a simple syntax summary, but, the reason I am posting this suggestion is because I don't understand the stuff just yet. Once I learn it (*if* I *can* learn it), I will gladly help manage the resource if needed.

The forum section would be straightforward, but probably not as efficient as some users might like - therein reducing the value to both the users and to Voxilla. Nevertheless, if you create a forum area specifically for the sharing of dial plans, I would offer to summarize and reformat the content on some regular basis if that would be valuable.

Thanks again for the great resource!

Randy

[PhoneBoy]

A general Sipura dial plan looks like (assume the numbers are subscripts)

(<a1:b1>c1<:@gwX1>|<a2:b2>c2<:@gwX2>|...)


<a:b> is basically a substitution syntax for stuff dialed at the beginning of the dial plan. OPTIONAL

Some examples include:

<8:1415> Replace a dialed "8" with "1415" (e.g. if user dials 84286511, we send 14154286511).

<:1415> Insert a "1415" into the number dialed

<9,:> When a 9 is dialed, present a second dialtone. The "9" is not sent.

<9:> If an initial 9 is dialed, don't send it.


c is a string of characters that restricts what digits can be dialed. THIS IS REQUIRED

These characters can be:

* means the star (asterisk) key

# means the pound (hash) key

x means a single digit (0 through 9)

[x-y] means any digits x through y.

[xy] means the digits x and y (can put any number of digits here)

. means repeat the previously listed digit type zero or more times. For example x. means "zero or more digits," xx. means "one or more digits."

S0 is usually used at the end of a dial plan string that says "immediately dial when matched." Used for things like 911.

! is used at the end of the dial plan and it means "immediately reject the number dialed." Used to block, say, 900 numbers.

A few examples:

[2-9]xxxxxx matches typical 7-digit dialing in the US.

011xx. matches North American style International dialing (011 followed by one or more digits)

[49]11S0 matches either 411 or 911 and immediately dials (doesn't wait for digit timeout)


<:@gwX> is specific to the SPA3000 and means "dial out this gateway." OPTIONAL

This is specific to the SPA3000 and is optional. Basically it means "if you match the dial plan listed, dial out this gateway." If nothing is specified in this part of the SPA-3000 dial plan on line 1, dial out via the VoIP provider defined on Line 1. The gateways are defined in the Gateway Accounts section of the Line 1 tab. gw0 is special and means "dial out the PSTN port."

gwX can also be replaced by a list of parameters that represent what's listed in the Gateway Accounts section. See the SPA Users Guide for more details.


Examples

Now for an example from the SPA3000 configuration Wizard, which uses most of these items

([2-79]11<:@gw0>|xx.|*xx.|**xx.|<#1,:>xx.<:@gw1>|<#9,:>xx .<:@gw0>|<#9,:>*xx<:@gw0>)

This breaks down as follows:

* 211, 311, 411, 511, 611, 711, and 911 are routed out the PSTN Line (gw0).
* An arbitrary number of digits, which will be routed out the Line 1 VoIP Provider configuration;
* dial * then dial one or more digits (routed out Line 1 VoIP Provider);
* dial ** then dial one or more digits (routed out Line 1 VoIP Provider);
* dial #1 then dial one or more digits (routed out Gateway 1 Provider (gw1));
* dial #9 then dial one or more digits (routed out PSTN Line);
* Dial #9* then two digits (routed out PSTN Line, i.e. for star codes).

Hopefully that's clear enough, but let me know if I'm missing something.

[bernart]

you left out blocking numbers and also making clear what part of the string is optional vs required

[PhoneBoy]

I assume you mean the ! syntax, correct? I'll go edit to annotate what's required.

[uo_mba]

The information below is from the Sipura FAQs. Of interest to some is the dialplan related to dialing specific IP addresses. The other info has already been discussed in Phoneboy's post.

**************************
A: The dialplan configuration is a combination of possible number dialing, separated by "|". Here are some examples:

• [2-9]xx xxxx 7-digit local number (cannot start with 0 or 1)
  
  1 xxx xxx xxxx US long distance, 1 + area code + local number
  
  011 xxxxxx x. US international + 6 or more digits
  
  <:1408> [2-9]xx xxxx adds "1408" to the 7 digit number that you dial
  
  <111:1002@10.10.10.2:5060> pressing 111, will dial the following IP address.

[rizsher]

I have the UK based Telappliant service, using it in the USA. So, all USA numbers become internatianal numbers, that too with 00 1 instead of the US Inetrnatial access code of 011. I tried the following modification to my Dial Plan under the Line 1 Tab:

(<:001>[2-9]xxxxxx|[2-79]11<:@gw0>|xx.|*xx.|**xx.|<#1,:>xx.<:@gw1>|<#9,:>xx .<:@gw0>|<#9,:>*xx<:@gw0>)

What I'm tryin to do is, to add 001 to the beginning of any number dialled startein with 2-9 plus another 6 digit, to allow 7 digit US dialling, e.g if I dial 217 555 5555, the Sipur should dial 001 217 555 5555; doesn't work. Where should this dial plan be, and is my DP correct?

Update: Fixed it with this:
(<:00>1[2-9]xx[2-9]xxxxxx|[2-79]11<:@gw0>|xx.|*xx.|**xx.|<#1,:>xx.<:@gw1>|<#9,:>xx .<:@gw0>|<#9,:>*xx<:@gw0>)

Next, want to simulate US dialling, i.e when I dial 011 44 208 xxxyyaa
i.e 011 followed by any number of digiguts, its should replace 011 with 00.

I tried:
(<:00>1[2-9]xx[2-9]xxxxxx|<011:00>xx x|[2-79]11<:@gw0>|xx.|*xx.|**xx.|<#1,:>xx.<:@gw1>|<#9,:>xx .<:@gw0>|<#9,:>*xx<:@gw0>)

Doesnt work.. any suggestion?

Corrected:

(<:00>1[2-9]xx[2-9]xxxxxx|<011:00>xxx.|[2-79]11<:@gw0>|xx.|*xx.|**xx.|<#1,:>xx.<:@gw1>|<#9,:>xx .<:@gw0>|<#9,:>*xx<:@gw0>)

[PhoneBoy]

The bolded part of your dial string should be:

<011:00>xxx.

i.e. you're missing a period.

[Kanak]

I noticed that you left out the 'P' option from your tutorial which I found very helpfull by the way.

I have the same problem as another person in the forum with a SPA3K behind a PBX.

I need to have a delay inserted mid stream when dialing a number from VoIP to PSTN to take into account the external line oppening zero.

I tried to use the pause option as described in the manual, but no luck. Here is what I tried:


(<0:0P5 >00011x.|[1-9]x)

I was hoping that this would let me dial the first zero, replace with a zero plus a pause, then dial the rest of the sequence (0011 in Australia is equivalent to 011 in the US).

I tried with a lagging blank and without. Same result, it ignores the pause.

The second part of the dial plan is simply to let me call an office extension from VoIP.

:?

[suebd]

I am trying to connect to an Asterisk system's voicemail, which is configured to be at 8500. When I dial 8500, I get the prompt to enter my mailbox number followed by the pound sign. But when I type in the mailbox followed by the pound sign, it seems to time out with a reorder signal.

I am able to dial in with a regular POTS line and retrieve voicemail, so the issue is with the Sipura 1000 or with something about how it is interacting with Asterisk. I have tried it with a regular corded phone and a cordless phone, and neither works, so it's unlikely it's the phone.

Here's my dial plan, which seems to work for everything else.

(*xx|[3469]11|0|00|xxxx|<:9>1xxx[2-9]xxxxxxS0|<:91314>[2-9]xxxxxxS0|xxxxxxxxxxxx.)

What's wrong? I've tried playing with the P (pause) value, to no avail...

[gbroiles]

Are you sure that the Sipura and the Asterisk box are using appropriate and compatible DTMF signalling settings, especially in light of whatever codec you've chosen? I have had trouble off-and-on with a similar issue - getting Asterisk to recognize DTMF generated by a phone on the other side of a Sipura 3000 - and finally got it working by getting the DTMF settings right. I think I settled on the RFC2822 (or similar) choice, but I'm on the wrong side of a NAT box right now to peek and see.