Because sipx supports both analog and digital gateways, you need to create a
custom dialing rule.
Dialed number
prefix (none) and "7" digits
Resulting call
Dial "1757" and append "Matched Suffix"
assign your sip trunk and restart any services required, place it "just above"
your existing local rule.
All the API does you are talking about is inject the xml rules for Asterisk.
sipx's dial plan allow you to create a dial plan that you need, and it
generates its own XML, so no API would be needed.
>>> "Andreas (Around the Clock Information Systems)" <andr...@atcis.net>
>>> 06/08/09 2:33 AM >>>
Thank you for your reply Mr. Picher,
Well I'm certainly not going to discount that as a possibility.
Before experimenting with sipXecs we were using trixbox CE which VoicePulse
supports directly. Basically, you just download a module written by
VoicePulse, install it on your trixbox server which enables some VoicePulse
API and then magically everything just starts working. It was so easy to
setup I couldn't believe it. If sipXecs catches on like I hope it will,
maybe VoicePulse will write a sipXecs compatible module too (he says with
fingers crossed). If any of you bad-a$$ programmers out there would like to
take a look at this API, here it is:
https://connect.voicepulse.com/secure/services/Api0605.asmx
I don't know much about programming, but it looks like it is mostly XML,
which I noticed there seems to be a lot of in sipXecs. If it makes sense to
any of you, perhaps VoicePulse would pay someone to port their code for them
to create an API module for sipXecs like they currently have for Asterisk
based IP PBX's.
Anyway, I kind of got sidetracked up there. I'll be contacting
VoicePulse technical support tomorrow to see what they have to say about it.
It's very possible that what Mr. Graziano had to say about it is correct
("VoicePulse probably expects you to send more than seven digits"), but
before I start barking up the wrong tree I'll verify it with them.
Thanks again to Mr. Picher and Graziano for your replies,
Andreas
Systems Engineer
Around the Clock Information Systems
-----Original Message-----
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Picher, Michael
Sent: Sunday, June 07, 2009 8:50 PM
To: Andreas (Around the Clock Information Systems);
sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Is there something wrong with my dial plan?
I'm not that familiar with the Cisco phones. Is there a Dial plan in
the phone that tells the phone when to send the dialed #?? It is
something we have to tweak with the Polycoms sometimes.
Mike
======================== Snip =============================
> local call for us is any number that begins with the Area Code of 757.
> Since my DID also begins with 757 I would think that I could pick a
> phone,
> dial seven (7) digits and my call would be connected. Not the case.
> Instead the phone (Cisco 7960) sometimes says "Proceeding (in 100)"
> followed
> by dead air indefinitely or after a few seconds simply says "Reorder"
> followed by a fast busy signal (this is what usually happens).
>
> My current Dial Rule for Local dialing is setup in a default
> manner
> except the check box for "Treat PSTN prefix as optional" and "Treat
> long
> distance prefix as optional" are both checked. I've tried toggling
> these
> check boxes both ways, but it doesn't seem to have any effect on seven
> digit
> dialing.
>
> So which is it? Do I have something configured incorrectly, or is
this
> an
> ITSP issue (VoicePulse)?
>
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