Configuring Asterisk for Skype Connect

Old Italian Phone

Old Italian Phone device

This post is intended to provide some basic guidance in configuring Skype Connect SIP Channels with Asterisk/Freepbx using SIP Trunk.

-- General Settings --
Outbound Caller ID: 99051000xxxxxx <Skype Connect registration ID>
Maximum Channels: 1 <The number of channels you purchased from Skype>

Dialed Number Manipulation Rules
(0039) + <prefix = NULL> | 0Z.
(0039) + <prefix = NULL> | 3Z.

-- Outgoing Settings --

Trunk Name: SkypeConnect <This can be whatever you want>

PEER Details:
username=99051000xxxxxx ; This is your Skype Connect registration ID
secret=xxxxxxxx ; This is your Skype Connect registration password
type=peer
qualify=yes
insecure=invite
host=sip.skype.com
fromdomain=sip.skype.com
disallow=all
allow=alaw&ulaw&g729
fromuser=99051000xxxxxx ; This is your Skype Connect registration ID

-- Registration --
Registration String: 99051000xxxxxx:<password>@sip.skype.com/99051000xxxxxx

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Strange issue with Sonicwall TZ 100/Asterisk 1.8 and a famous VOIP ISP

I want to tell a very strange issue occoured some weeks ago using pbx Asterisk 1.8, an router/firewall Sonicwall TZ 100 and a famous Voip ISP.

The system has always worked fine until a couple of weeks ago (pls see in linkografia for config used in Sonicwall TZ100).

Without any warning one day calls began to be randomly “one way”: the caller can hear but the called cannot. Briefly here is what we have verified togheter with the Voip ISP customer care.

Asterik & SonicWall & Voip ISP
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Configure a LinkSys SPA942 with Asterisk 1.8

In this post I will show you how to connect & configure one IpPhone Linksys SPA942 using an Asterisk Pbx 1.8: I hope this can help you with other similar IpPhones (Linksys SPA942 is in “End-of-sale “)

To start you have to verify the firmware version used by the phone: connect to the web interface on the IpPhone and check the info tab check.

LinkSys SPA942 - Firmware version
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