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Cellphone Codes

2012-05-16T12:41:35Z

Move Current Call to Cell Phone (*52):

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	== Retrieve Call From Cell Phone (*51) ==		== Retrieve Call From Cell Phone (*51) ==

-	This feature allows ~~users~~ to pick up an in-process cell phone call ~~on their extension~~ phone. This feature can be used only for calls that have been made from the cell phone (through the ~~Personal Virtual Assistant~~) or which have been redirected to the user’s cell phone. Calls that have been made directly to the user’s cell phone (without involving the system) cannot be picked up on the user’s ~~extension~~ phone.	+	This feature allows you to pick up an in-process cell phone call from your office phone. This feature can be used only for calls that have been made from the cell phone (through the PBX) or which have been redirected to the user’s cell phone. Calls that have been made directly to the user’s cell phone (without involving the system) cannot be picked up from the user’s office phone.

-	''Scenario'': You’re on your way to the office and you need to call a client, so you call into the system and place an outbound call. You speak with the client on your way to work, and once you arrive at the office, you decide to pick the call up from your office phone.	+	''Scenario'': You’re on your way to the office and you need to call a client, so you call into the system and place an outbound call. You speak with the client on your way to work, and once you arrive at the office, you decide to pick the call up from your office phone. Here's what you would do:
-		+
-	'~~''To transfer a cell phone call to your extension~~:~~'''~~	+

	#From your cell phone, call into the main number of the PBX (your cell phone must be associated with your account).		#From your cell phone, call into the main number of the PBX (your cell phone must be associated with your account).
-	#~~From the Personal Virtual Assistant menu, press~~ 1 ~~for “outbound call~~.”	+	#Press 1 to dial a number.
	#Enter a destination number, then press the # key.		#Enter a destination number, then press the # key.
-	#When ~~you’re~~ ready to ~~pick~~ the call ~~up from~~ your office phone, dial *51 from the office phone.	+	#When you're ready to transfer the call to your office phone, just dial *51 from the office phone.

	Once the call has been connected, your cell phone will disconnect and will no longer accumulate minutes.		Once the call has been connected, your cell phone will disconnect and will no longer accumulate minutes.
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	== Move Current Call to Cell Phone (*52) ==		== Move Current Call to Cell Phone (*52) ==

-	This feature allows ~~users~~ to move a current call from the ~~extension~~ phone to the cell phone. (''Note'': The user’s cell phone must be configured to the extension before this feature can be used.)	+	This feature allows you to move a current call from the office phone to the cell phone. (''Note'': The user’s cell phone must be configured to the extension before this feature can be used.)

-	''Scenario'': You’re ~~on your extension~~ in the middle of a conversation and need to leave the office. Instead of ending the conversation, you put the caller on hold and transfer the call to your cell phone.	+	''Scenario'': You’re in the middle of a conversation on your office phone and need to leave the office. Instead of ending the conversation, you put the caller on hold and transfer the call to your cell phone.

-	'''To transfer ~~an extension~~ call to your cell phone:'''	+	'''To transfer a call from your office phone to your cell phone:'''
-	#~~Place~~ the ~~extension~~ call on hold.	+	#From your office phone, place the call on hold.
	#Dial *52 from your office phone.		#Dial *52 from your office phone.
	#Answer the call on your cell phone.		#Answer the call on your cell phone.

Main Page

2012-05-15T09:29:32Z

Other Products:

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	<table border="0" cellpadding="5" cellspacing="5" width="100%">		<table border="0" cellpadding="5" cellspacing="5" width="100%">
	<tr>		<tr>
-	<td width="10%">[[File:SoHo_small.gif]]</td><td width="10%">[[File:plus_icon.jpg]]</td><td width="10%">[[File:ICUsmall.jpg]]</td><td width="10%">~~[[File:mini_icon.png]]~~</td>	+	<td width="10%">[[File:SoHo_small.gif]]</td><td width="10%">[[File:plus_icon.jpg]]</td><td width="10%">[[File:ICUsmall.jpg]]</td><td width="10%"></td>
	</tr>		</tr>
	<tr>		<tr>
-	<td width="12%">[[The_snom_ONE_SoHo\|snom ONE SoHo]]</td><td width="12%">[[The_snom_ONE_plus\|snom ONE plus]]</td><td width="17%">[[The_snom_ICU_doorkeeper\|snom ICU Doorkeeper]]</td><td width="10%">~~[[The_snom_ONE_mini\|snom ONE mini]]~~</td>	+	<td width="12%">[[The_snom_ONE_SoHo\|snom ONE SoHo]]</td><td width="12%">[[The_snom_ONE_plus\|snom ONE plus]]</td><td width="17%">[[The_snom_ICU_doorkeeper\|snom ICU Doorkeeper]]</td><td width="10%"></td>
	</tr>		</tr>
	</table>		</table>

System SIP Settings

2012-05-14T15:40:31Z

SIP Settings:

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	*'''Loopback detection''': This setting applies only to multi-domain environments. Single-domain environments are required to leave this setting on.		*'''Loopback detection''': This setting applies only to multi-domain environments. Single-domain environments are required to leave this setting on.

		+	<!--
	*'''Inband DTMF detection''': When a user presses a key on the telephone, the system must be able to understand the action. In telephony systems, this mechanism is typically called DTMF. In VoIP, DTMF is usually sent “out of band” (RFC 2833) because it allows the system to easily detect the tones in a fail-safe manner. However, not all devices support this method, and when this is the case, the system must decode and analyze the media stream and perform this detection. Because this is erroneous and costs additional CPU performance, it is strongly recommended that you not use this feature and instead replace devices which do not support the out-of-band method with devices that do. In-band and out-of-band can be compared to a T1 line that uses inband signalling and a PRI line that uses out-of-band signalling.		*'''Inband DTMF detection''': When a user presses a key on the telephone, the system must be able to understand the action. In telephony systems, this mechanism is typically called DTMF. In VoIP, DTMF is usually sent “out of band” (RFC 2833) because it allows the system to easily detect the tones in a fail-safe manner. However, not all devices support this method, and when this is the case, the system must decode and analyze the media stream and perform this detection. Because this is erroneous and costs additional CPU performance, it is strongly recommended that you not use this feature and instead replace devices which do not support the out-of-band method with devices that do. In-band and out-of-band can be compared to a T1 line that uses inband signalling and a PRI line that uses out-of-band signalling.
-		+	-->
	*'''Remote SIP management''': This setting allows the provider to send commands to the system (for example, for rereading the configuration). It is useful in environments where the service provider controls the system from a centralized location. By default, this setting is off, but if you are using such an environment, this setting needs to be activated.		*'''Remote SIP management''': This setting allows the provider to send commands to the system (for example, for rereading the configuration). It is useful in environments where the service provider controls the system from a centralized location. By default, this setting is off, but if you are using such an environment, this setting needs to be activated.

So Tel Systems Trunk Settings

2012-05-14T11:09:05Z

SIP Trace:

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		+	== Trunk Settings ==
		+
	[[File:sotel1.jpg]]<br>		[[File:sotel1.jpg]]<br>
	<br>		<br>
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	<br>		<br>
	[[File:media9.png]]		[[File:media9.png]]
		+
		+	== SIP Trace ==
		+
		+	[5] 2012/05/11 14:06:53: SIP Tx udp:4.56.160.134:5060:
		+	<br>
		+	INVITE sip:13234276156@register2.sotelips.net;user=phone SIP/2.0
		+	<br>
		+	Via: SIP/2.0/UDP 74.142.110.2:5060;branch=z9hG4bK-5b0dd808d44bbf6d80cc89806e6dd7cd;rport
		+	<br>
		+	From: "18591234567" <sip:18591234567@register2.sotelips.net>;tag=30345
		+	<br>
		+	To: <sip:13234276156@pbx.civic-finance.com;user=phone>
		+	<br>
		+	Call-ID: 6dafa742@pbx
		+	<br>
		+	CSeq: 17682 INVITE
		+	<br>
		+	Max-Forwards: 70
		+	<br>
		+	Contact: <sip:18591234567@71.142.115.2:5060;transport=udp>
		+	<br>
		+	Supported: 100rel, replaces, norefersub
		+	<br>
		+	Allow-Events: refer
		+	<br>
		+	Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
		+	<br>
		+	Accept: application/sdp
		+	<br>
		+	User-Agent: snomONE/4.5.0.1075 Delta Aurigids
		+	<br>
		+	P-Asserted-Identity: "18591234567" <sip:18591234567@register2.sotelips.net>
		+	Content-Type: application/sdp