Voipbuster Setup Instructions for Linksys PAP2

Sample configuration of Linksys PAP2 for Voipbuster SIP service provider.

Voipbuster.com

1) Follow unlock instructions above.
2) Log into your adapter, click on ADMIN link, click on Advanced View link, go to Line 1 tab (or Line 2) and enter the following values in each parameter:

Line enable : yes
nat mapping enable : yes
proxy : stun.voipbuster.com
outbound proxy : nat.voiptalk.org:5065
use outbound proxy : yes
register : yes
display name : voipbuster username
user id : voipbuster username
password : voipbuster password


For USA residents, add the following line to Dial Plan field:
(<:001aaa>[2-9]xxxxxx|<:00>1[2-9]xx[2-9]xxxxxxS0|<:001>[2-9]xx[2-9]xxxxxxS0|<011:00>xx.)

Replace aaa with your area code.

This dial plan handles three types of calling:

Local : xxx-xxxx or xxx xxx-xxxx
Long distance : 1 xxx xxx-xxxx or xxx xxx-xxxx
International : 011 + country code + area code + number
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VBuzzer Setup Instructions for Linksys PAP2

Sample configuration of Linksys PAP2 for VBuzzer SIP service provider.

VBuzzer.com

Line Enable: YES
SAS Enable: NO
SIP Port: 5080
Proxy: vbuzzer.com
Outbound Proxy: vbuzzer.com:80
Use Outbound Proxy: YES
User ID: (my vbuzzer user name)
Password: (my vbuzzer PW)
Dial Plan: (*xx|[3469]11|0|00|[2-9]xx xxxx|[0]xxx xxx xxxx|xxxxxxxxxx|xx xxx xxx xxxx.|*xx*xx.)
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Stanaphone Setup Instructions for Linksys PAP2

Sample configuration of Linksys PAP2 for Stanaphone SIP service provider

Stanaphone.com

1) Follow unlock instructions above.
2) Log into your adapter, click on ADMIN link, click on Advanced View link, go to Line 1 tab (or Line 2) and enter the following values in each parameter:

Line enable : yes
nat mapping enable : yes
Username : XXXXXXXXXX (use your StanaPhone telephone number)
Authorization User : XXXXXXXXXX (use your StanaPhone telephone number)
Password : UUUUUUUUUUUU (use your Stanaphone password - different from your StanaPhone account password - login to Stanaphone and click Account Information link, then Config SIP link to get this)
Domain/Realm SIP Proxy : sip.stanaphone.com
Outbound Proxy : (leave empty)
DTMF : RFC-2833

Dial Plan: (<:1aaa>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|<:1>[2-9]xx[2-9]xxxxxxS0|011xx.)

Replace aaa with your area code.

This dial plan handles three types of calling:

Local : xxx-xxxx or xxx xxx-xxxx
Long distance : 1 xxx xxx-xxxx or xxx xxx-xxxx
International : 011 + country code + area code + number

Alternate Dial Plan: (08xxxxxxS0|<:1AAA>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|<:1>[2-9]xx[2-9]xxxxxxS0|011xx.)

Adds Stana-to-Stana : 08xxxxxx

Other parameters use default values.

SIP tab:

Handle VIA received: Yes Handle VIA rport: Yes
Insert VIA received: Yes Insert VIA rport: Yes

STUN Enable: Yes
STUN Server: stun.fwdnet.net:3478

Line 1 Tab:

NAT Mapping Enable: Yes
SIP Port: 5061
Proxy: sip.stanaphone.com
Register Expires: 400
Preferred Codec: G.711a
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FreeWorldDialup Setup Instructions for Linksys PAP2

Sample configuration of Linksys PAP2 for FreeWorldDialup voip provider.

freeworlddialup.com

Line Enable: YES
NAT Mapping Enable: YES; NAT Keep Alive Enable: YES
SIP Port: 5061

Proxy: fwd.pulver.com; Use Outbound Proxy: YES

Outbound Proxy: fwdnat.pulver.com:5082

Display Name: (Your name); User ID: (Your FWD number)
Password: (Your FWD password)

Preferred Codec: You must choose 711u
Use Pref Codec Only: NO

Dial Plan: (xxx|xxxx|xxxxx|xxxxxx|1xxxxxxxxxx|*1xxxxxxxxxx|**1xxxxxxxxxx|*xx*xx.)
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What is SIP?

SIP, the session intitation protocol, is the IETF protocol for VOIP and other text and multimedia sessions, like instant messaging, video, online games and other services.

Abstract from the SIP RFC 3261 (latest formatted/explained version)

This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.

SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.

SIP is very much like HTTP, the Web protocol, or SMTP. Messages consist of headers and a message body. SIP message bodies for phone calls are defined in SDP -the session description protocol.

• SIP is a text-based protocol that uses UTF-8 encoding
• SIP uses port 5060 both for UDP and TCP. SIP may use other transports

SIP offers all potentialities of the common Internet Telephony features like:

• call or media transfer
• call conference
• call hold

Since SIP is a flexible protocol, it is possible to add more features and keep downward interoperability.

SIP also does suffer from NAT or firewall restrictions. (Refer to NAT and VOIP)

SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP) services. The following features of SIP play a major role in the enablement of IP telephony and VoIP:

• Name Translation and User Location: Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
• Feature Negotiation: This allows the group involved in a call (this may be a multi-party call) to agree on the features supported � recognizing that not all the parties can support the same level of features. For example video may or may not be supported; as any form of MIME type is supported by SIP, there is plenty of scope for negotiation. Call Participant Management - During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
• Call feature changes: A user should be able to change the call characteristics during the course of the call. For example, a call may have been set up as �voice-only�, but in the course of the call, the users may need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call
• Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call, enable selection of the appropriate codec for establishing a call between the various devices. This way, less advanced devices can participate in the call, provided the appropriate codec is selected.

Source
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Welcome to Dot Sip

This blog is dedicated toward Voip & Sip related information.

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