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jomaha Offline OP
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I have just installed a RPTU1. I'm having trouble routing incoming calls to 4 digit extensions. To test I set up a 3 digit extension in program *09 as I remembered the default length is 3 - and the incoming call worked fine.

I have
*64-1 LED1 'on' for the trunk group (06)
*64-2 Set to '4' on this trunk group

Something appears to be trumping the *64-2 setting. Any ideas?

Thanks,

John

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Check program 17.

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jomaha Offline OP
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I have tried various settings for program 17 - now all LEDs are off. But specifically LEDs 7&5 are off. At any rate, the *09 works for 3 digit definitions.

Additionally, I should have mentioned that the failed calls are arriving to the automated attendant via hunt defined in *17. What else? I tried entering a value in *66-5 and could not enter a 'wild card' with either '*' or LED2 although I was able to enter 7-blank-blank-blank the documentation on this program is a bit unclear on this - but again the 3 digit dialing works which suggests that I shouldn't need a value defined here - for that matter, entering an actual 4 digit extension in this field doesn't help either.

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Wait a minute. Do you have 3 digits extensions or 4 digit extensions or both? phonemaster may have some understanding, but I am lost as to what you are doing.

Any DID calls that don't have a valid destination will go to the attendant by default, to another extension at your choice.


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Are you saying that when telco sends you 4-digit DIDs it fails? Or are you saying that when you change a port assignment in prog *09 to a 4-digit DID it fails? If you cannot enter an asterick in program *66-5 then your software level is below 4.15 and should be upgraded. Also, the RPTU1 should be an RPTU1B with the latest rom chips on it. The RPTU1A had problems and could not be upgraded.

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jomaha Offline OP
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Maybe I should explain what I am connected to. I do the phones for a school district with several non-contiguous locations. I am using asterisk to host sip phones at some buildings, at others, I am using asterisk to bridge to legacy pbxs (Merlin legends, sys75... some of these are being phased out in favor of the sip) with PRI or a regular tie T (current connection to the DK424). This all works great and has enabled us to get rid of a real lot of telco service but I want caller ID going in and out of the Toshiba and I want to move auto-attendant & voicemail to asterisk.

The non toshiba dial plan consists of 6XXX (currently reached by dialing 9-6XXX) extensions. All extensions within the toshiba are of the form 7XXX. Anyway, for the purposes of this discussion, you can see our system as...
telco---pri---asterisk---pri---dk424
...on asterisk I have a great deal of flexibility to configure the T as a variety of PRI flavors, network-or-customer-end... so you can view the asterisk box as the telco - but I have complete control of it.

Now to address your questions, program 00-0 says my ver=REA4X2N. Is it straightforward to upgrade (if necessary) and where can I obtain a newer binary?
I may be stuck with the RPTU1A. Are the issues with particular PRI switchtypes?

To touch Bunnie's post, (with the exception of interal stuff like phantom extensions) all toshiba phones have 4 digit extensions which I can't get to (regardless of *09) - but from asterisk I can send whatever I want into the toshiba and that is how I found I could reach the 3 digit extensions defined in *09.

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OK, you have 3 digit phantom extensions and 4 digit actual phone extensions and your sip phones can get to phantoms but not actual phones correct?

(I really am trying to work on this.)

From what, from where can't you get to the 4 digit extensions that you can reach with Asterisk.

You are way ahead of me on Asterisk but not clearly defining, at least to me, the exact flow that doesn't flow. I was a math major, but not an English major.


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In *09 you should be associating a Port number on the Toshiba with a 3 or 4 digit DID number being sent to the Toshiba from the Asterisk. The port number will be assigned to an extension on the Toshiba in Program 04. So if you are sending from the Asterisk digits 6000 and you wanted it to go to extension 7000 you would have to see what port 7000 extension is assigned to in program 04
Program *09 = 000 # 6000 or 600 depending on digits your are sending from Asterisk
Program 04 Port ??? = 7000
It sound like in your post you were trying to assign a Toshiba extension in *09 and you have to assign the port the extension is assigned to.

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Have you tried using program 71? It's more flexible than *09.

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jomaha Offline OP
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>>OK, you have 3 digit phantom extensions and 4 digit actual phone extensions and your sip phones can get to phantoms but not actual phones correct?

Not exactly, if for instance I map
port 000 to 123 in pgm *09 I can now ring 'the actual extension at port 000' by dialing in 123 from asterisk.

I'm just trying pgm 71, I was able to get certain 4 digit extensions through (pri calls originating in the toshiba) but the issue is the caller - this may be part of the RPTU1A problem - but I'm halfway there!

Thanks,

John

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