Mostrando entradas con la etiqueta SIP. Mostrar todas las entradas
Mostrando entradas con la etiqueta SIP. Mostrar todas las entradas

lunes, 5 de mayo de 2008

DUNDi Enterprise Configuration SIP

This is all you should need to setup DUNDi between two boxes with SIP.
You can add more boxes to your network in a similar way.

(Based on configurations in the configs directory of the Asterisk source code and documents published by Brian K. West aka bkw_)
DUNDi Enterprise Configuration IAX

Note: This configuration is currently non-functional, as chan_sip does not support "dbsecret" at this time. You can create a configuration where the passwords are actually included in the DUNDi mappings or in the peer definitions. A patch to support dbsecret in chan_sip is forthcoming (for CVS HEAD only).

Things to keep in mind:
  • Unless you have specifically listed a host in your sip.conf the call will come in on the context defined in the [general] section by context. You may want to include the [dundi-priv-local] in this context.


On both boxes in your extensions.conf:
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers

[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.

[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here

[dundi-priv-local]
include => dundi-priv-canonical
include => dundi-priv-customers
include => dundi-priv-via-pstn

[dundi-priv-switch]
; Just a wrapper for the switch
switch => DUNDi/priv

[dundi-priv-lookup]
include => dundi-priv-local
include => dundi-priv-switch

[macro-dundi-priv]
exten => s,1,Goto(${ARG1},1)
include => dundi-priv-lookup


sip.conf on both boxes:
[priv]
type=user
dbsecret=dundi/secret
context=dundi-priv-local


dundi.conf on both boxes under mappings:
In many cases you will need to replace ${IPADDR} with your local IP address
priv => dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial



now on each box cd /var/lib/asterisk/keys

astgenkey -n BOXNAMEHERE

Press enter do not put a password on the keys unless you want to init
keys every time you start asterisk.

Now put exchange public keys between the boxes.



Box A dundi.conf:
[DE:AD:BE:EF:DE:AD]   <-- EID/MAC from BOX B model = symmetric host = boxb.domain.com inkey = BOXB   <- BOX B's public key outkey = BOXA  <- BOX A's private key include = priv permit = priv qualify = yes order = primary 


Box B dundi.conf:

[BE:EF:DE:AD:BE:EF]  <-- EID/MAC from BOX A model = symmetric host = boxa.domain.com inkey = BOXA   <- BOX A's public key outkey = BOXB  <- BOX B's private key include = priv permit = priv qualify = yes order = primary 


Now you can do this in the context which your devices dialout:
exten => _91NXXNXXXXXX,1,Macro(dundi-priv,${EXTEN:1})
exten => _91NXXNXXXXXX,2,Dial(Zap/g1/${EXTEN:1}) ; This is fall through example to a PSTN such a as PRI


And find numbers in your enterprise DUNDi network.

These same things apply to the e164 DUNDi network too.


http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP


DUNDi Enterprise Configuration SIP with no passwords

http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords


DUNDi Enterprise Configuration IAX

http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+IAX

miércoles, 28 de noviembre de 2007

Firewall, NAT y SIP

Sip ,trunking, nat y los firewall...los cuales no son faciles.. (http://www.ingate.com/)

Ingate SIParators

http://www.ingate.com/SIP-trunking.php

he most common solution for enterprises with a local IP-PBX to connect to the PSTN is via a local PSTN gateway. However, today more and more service providers offer one single connection for both data and voice between the carrier and the enterprise. This is commonly called a SIP trunk. For the enterprise this means that they no longer need either the local PSTN gateway or costly PRI/BRI?s as the local IP-PBX can be connected to the service provider?s PSTN gateways over the Internet.

The SIP Trunking software module, working in conjunction with an Ingate Firewall or SIParator, solves the Network Address Translation (NAT) traversal issues that are faced by businesses using a SIP trunk. Together, they control both incoming and outgoing communications and route the communication to the intended users. All voice traffic (as well as data traffic) must traverse the enterprise firewall/NAT. However, SIP traffic cannot traverse traditional enterprise firewalls and NAT devices. As a result, the firewall/NAT device blocks all SIP traffic, which includes VoIP. Ingate resolves this issue, enabling enterprises to utilize SIP trunks and enjoy the benefits of SIP trunking while maintaining the security and integrity of their network.
A cost-efficient solution, the SIP trunk reduces monthly expenses for the enterprise as they:

* Only need one connection for both data and voice
* No longer need PRI/BRI connections
* Don?t have to buy or maintain a local PSTN-gateway

Firewalls and Benefits of Proxy-Based Solutions -- Technical details

The Ingate Firewall® or Ingate SIParator® solves the basic problem of firewall and NAT traversal using a SIP proxy-based implementation. The SIP proxy is a complete solution to the firewall and NAT traversal issues introduced by the enterprise firewall. A proxy is designed to briefly stop the signaling packets so that each one can be inspected before the header information is rewritten. The SIP proxy also makes sure the packets are delivered to the appropriate endpoints. This provides the enterprise with a flexible, controlled implementation of SIP-based communications that also maintains voice/video quality. Ingate Firewalls and SIParators offer a number of advantages:

* Near-end NAT traversal for connecting SIP communications between an enterprise and a carrier
* Dynamically open and close the media ports to admit voice or video on both UDP and TCP ports
* Encryption for communication privacy and prevention of eavesdropping. Include termination, transcoding and passthrough of TLS and SRTP
* Authentication and registration at the carrier side
* Give priority of voice traffic using the Ingate Quality of Service features
* ENUM look-up to route the call IP-IP all the way if possible
* Least cost routing of international calls

Confirmed interoperability

Ingate has confirmed interoperability several of the leading IP-PBX vendors and ITSPs. See the full list here.
Startup Tool for SIP Trunking

The Startup Tool is an installation tool for Ingate Firewall® and Ingate SIParator® products using the Ingate SIP Trunking software module, which facilitates the setup of complete SIP trunking solutions. The tool will automatically configure a user?s Ingate Firewall or SIParator to work with the IP-PBX or SIP trunking service provider of their choice. With the push of a button, the configuration tool will automatically create a SIP trunk connection designed to the user?s individual setup.
Read more about the tool here.

Read more about the Ingate SIP Trunking software module

Learn more about the Ingate Firewall

Learn more about the Ingate SIParator

For additional information on Ingate SIP trunking solution please contact info@ingate.com

lunes, 24 de septiembre de 2007

Helius SIP

on este tipo de documentacion es la que hay que trabajar desarrollando...ya que existen muchas cosas en los protocolos..que no estan siendo usadas o subaprovechadas...
RFC 3261 - SIP: Session Initiation Protocol http://www.faqs.org/rfcs/rfc3261.html