Dialing Out

I have the latest version of FreePBX and bunch of Polycom 501 handsets. The system is configured to select the proper trunk to dial out, and it does if I punch in the numbers and then hit the dial button on the phone. However, if I pickup the handset, or put it on speakerphone and dial the same digits, it errors out saying that the call cannot be completed as dialed. It acts as if it follows a different dial pattern when the second method of dialing is used. I have looked all over the place to find where the second dial pattern is stored and NO luck. Any insight on this is greatly appreciated!

This is the phones digit map. It matches the internal dial plan so users don’t have to press the send key.

How you configure it depends on your provisioning method. If you are using a tftp configuration server the dial plan is in sip.cfg. You may also be able to access it from the phones web configuration page. I don’t recommend that method other than for testing.

On a Polycom when you “cold dial” as you have described the digits bypass the digit map. Whenever you hear dial tone “hot dialing” the dial plan is in effect.

A properly crafted dial plan will allow you to dial internal extensions, local and LD without pressing the send key.

SkykingOH, you helped me on another issue I had with making the phones handle multiple calls. THANK YOU! I am sure you don’t remember details. As a refresher, I am very new to this system, and for now I need a bit more hand holding. I am studying the system in detail however, and am eager to learn more!
Are there any documentations on how to craft a dial plan. I have the Polycom manual, but I don’t see anything in there about hot or cold dial concept you mentioned. I am using tftp, and looked at the sip.cfg. Since I am not familiar with the programming language used to create sip.cfg, I cannot follow the logic. Are there any documentations/tutorials on that? If not, perhaps you can elaborate a bit more on what to do. Thanks again!

You have the system administrators guide for the SIP release you are running?

The sip.cfg is XML, if you copy the file to your Windows workstation (I use the fee tool WinSCP) you can then edit with XML Notepad, a free Microsoft application that makes it easy to edit XML.

I looked at the document titled “Administrator’s Guide for the
Polycom® SoundPoint®
IP/SoundStation® IP/ VVX™

ANd on page A-25 found the documentation for the Digit Map variable along with two pages of examples.

Thanks pal, I was looking at the the wrong book thinking this is a programing issue. Will keep you posted if I get stuck. Thanks again!