Atcom AG188N

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[draft revision]

[draft revision]

Revision as of 19:26, 21 January 2012 (view source) Carlb (Talk | contribs) m (+cat) ← Older edit		Revision as of 21:31, 28 October 2013 (view source) Jeff (Talk | contribs) (→Configuration Details) Newer edit →
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	==Configuration Details==		==Configuration Details==
		+	__TOC__
-	'''SIP Configuration Section:'''	+	=='''SIP Configuration Section:'''==
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-	[[File:Atcom.JPG|Atcom AG188N Configuration Screen|frame|left]]	+	[[File:Atcom.JPG|Atcom AG188N Configuration Screen]]
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		+	==Known Issues with this Device:==
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		+	'''Problems when dialing *97 for Voicemail:'''
		+
		+	With one phone hooked to the ‘phone’ jack, you can dial a star code to access the PSTN line instead of the SIP/IAX configured on the ATA. So, one analog phone enables you to use either SIP/IAX or the PSTN. However, people using the ATA with Asterisk have a problem when they want to dial Asterisk * codes like *97 for voicemail (as the * takes you to PSTN). The solution is to delete the “lifeline *T” option from “Dial-Peer” menu.
	[[category:Analog Telephone Adapters]]		[[category:Analog Telephone Adapters]]

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Revision as of 21:31, 28 October 2013

Configuration Details

Contents 1 Configuration Details 2 SIP Configuration Section: 3 Known Issues with this Device:

SIP Configuration Section:

• Fill the following fields according to your account:
  • Register Server Addr: atlanta.voip.ms (one of our multiple servers)
  • Register Server Port: 5060
  • Register Username: 100000 (your VoIP.ms username)
  • Register Password: ******** (the account password)
  • Detect Interval Time: 60
  • DTMF Mode: DTMF_RFC2833
  • Enable Via rport: Enabled
  • Local SIP Port: 5060
  • Register Expire Time: 60
  • RFC Protocol Edition: RFC3261
  • Server Type: common
  • SIP Default Protocol: Enabled

• Click on the "Apply" button at the bottom of the form.

Configuration Screen

Known Issues with this Device:

Problems when dialing *97 for Voicemail:

With one phone hooked to the ‘phone’ jack, you can dial a star code to access the PSTN line instead of the SIP/IAX configured on the ATA. So, one analog phone enables you to use either SIP/IAX or the PSTN. However, people using the ATA with Asterisk have a problem when they want to dial Asterisk * codes like *97 for voicemail (as the * takes you to PSTN). The solution is to delete the “lifeline *T” option from “Dial-Peer” menu.