2 - SIP y H.323: Establecimiento y gestión de sesiones multimedia Asterisk

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2 -V

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IP (V IP)

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IP (V IP)

Voz sobre IP (VoIP)

SIP y H.323: Establecimiento y

gestión de sesiones multimedia _{Computer Networking: A}

Asterisk

Computer Networking: A Top Down Approach Featuring the Internet, 3rd _edition.

Jim Kurose, Keith Ross Addison Wesley July 2004 Addison-Wesley, July 2004.

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C 2011-2012

Voice-over-Data (VoD) Enables New Applications

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 “Click to talk” web sites for e-commerce

 Digital white-board conferences

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AS  Digital white board conferences

 Broadcast audio and video over the Internet or a

corporate Intranet

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 Integrated messaging: check (or leave) voice mail over

the Internet TECNOL O G  Instant messaging  Voicemail notifications  Stock notifications  Callback notification  Fax over IP  Fax over IP  Etc. 2

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C 2011-2012

Sesion Initiation Protocol

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 SIP is end-to-end, client-server session signaling

protocol D AVA N ZA D AS

 SIP’s primarily provides presence and mobility

 Protocol primitives: Session setup, termination, changes,...

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 Arbitrary services built on top of SIP, e.g.:

 Redirect calls from unknown callers to secretary  Reply with a webpage if unavailable

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 Send a JPEG on invitation

 Features:

 Textual encoding (telnet, tcpdump compatible).  Programmability.g y

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C 2011-2012 Where’s SIP S –M as te r IC D AVA N ZA D AS SDP codecs G ÍAS DE RE D Application RTSP SIP SDP codecs RTP DNS(SRV) TECNOL O G Transport TCP UDP Network _IP

Physical/Data Link _Ethernet

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C 2011-2012

IP SIP Phones and Adaptors

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 Are true Internet

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Analog phone adaptor  Are true Internet

hosts  Choice of application G ÍAS DE RE D 3  Choice of application  Choice of server  IP appliances TECNOL O G 3 Palm control  Implementations  3Com (3) control  Columbia University  MCI WorldCom (2)  Mediatrix (1)  Mediatrix (1)

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C 2011-2012 SIP Components S –M as te r IC  User Agents

 UAC (user agent client): Caller application that initiates and sends SIP requests.  UAS (user agent server): Receives and responds to SIP requests on behalf of

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AS  UAS (user agent server): Receives and responds to SIP requests on behalf of clients; accepts, redirects or refuses calls.

 Server types

 Redirect Server

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 Accepts SIP requests, maps the address into zero or more new addresses and returns those addresses to the client. Does not initiate SIP requests or accept calls.

 Proxy Server

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 Contacts one or more clients or next-hop servers and passes the call requests further. Contains UAC and UAS.

 Registrar Server

 A registrar is a server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles.

 Location Server

 Provides information about a caller's possible locations to redirect and proxy servers

 Provides information about a caller s possible locations to redirect and proxy servers. May be co-located with a SIP server.

 Gateways

 A Sip Gateway service allows you to call 'real' numbers from your software or

 A Sip Gateway service allows you to call real numbers from your software or have a dedicated 'real' telephone number which comes in via VoIP

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C 2011-2012 SIP Trapezoid S –M as te r IC DNS Location D AVA N ZA D AS DNS Server Location Server DNS Registrar G ÍAS DE RE D Outgoing Proxy SIP Registrar Incoming Proxy TECNOL O G y SIP SIP y SIP Terminating Originating SIP

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C 2011-2012 SIP Triangle? S –M as te r IC DNS Location D AVA N ZA D AS DNS Server Location Server DNS Registrar G ÍAS DE RE D Registrar Incoming Proxy TECNOL O G SIP SIP y SIP Terminating U A Originating User Agent SIP RTP _{User Agent} User Agent RTP 8

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C 2011-2012

SIP Peer to Peer!

S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G Terminating Originating SIP

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C 2011-2012 SIP Methods S –M as te r IC

 INVITE Requests a session

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AS  INVITE Requests a session

 ACK Final response to the INVITE

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 OPTIONS Ask for server capabilities

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 CANCEL Cancels a pending request

 BYE Terminates a session

 REGISTER Sends user’s address to server

 REGISTER Sends user s address to server

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C 2011-2012 SIP Responses S –M as te r IC  1XX Provisional 100 Trying D AVA N ZA D AS  2XX Successful 200 OK G ÍAS DE RE D

 3XX Redirection 302 Moved Temporarily

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 4XX Client Error 404 Not Found

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C 2011-2012

SIP Flows - Basic

S –M as te r IC User A User B D AVA N ZA D AS INVITE: sip:18.18.2.4 “Calls” 18.18.2.4 180 - Ringing Rings G ÍAS DE RE D ACK 200 - OK Answers TECNOL O G RTP Talking Talking 200 - OK BYE Hangs up 1 2

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C 2011-2012 SIP INVITE S –M as te r IC

INVITE sip:e9-airport.mit.edu SIP/2.0

From: "Dennis Baron"<sip:[email protected]>;tag=1c41

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AS From: Dennis Baron <sip:[email protected]>;tag 1c41

To: sip:e9-airport.mit.edu

Call-Id: [email protected]

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Contact: "Dennis Baron"<sip:[email protected]> Content-Type: application/sdp TECNOL O G Content-Length: 304 Accept-Language: en

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Supported: sip-cc, sip-cc-01, timer, replaces

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C 2011-2012

Session Description Protocol

S –M as te r IC  IETF RFC 2327

 “SDP is intended for describing multimedia sessions for

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AS  SDP is intended for describing multimedia sessions for

the purposes of session announcement, session invitation, and other forms of multimedia session

G ÍAS DE RE D initiation.”  SDP includes: TECNOL O G

 The type of media (video, audio, etc.)

 The transport protocol (RTP/UDP/IP, H.320, etc.)

 The format of the media (H 261 video MPEG video etc )  The format of the media (H.261 video, MPEG video, etc.)

 Information to receive those media (addresses, ports, formats

and so on)

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C 2011-2012 SDP S –M as te r IC v=0 D AVA N ZA D AS v 0 o=Pingtel 5 5 IN IP4 18.10.0.79 s=phone-call G ÍAS DE RE D _{c=IN IP4 18.10.0.79} t=0 0 m=audio 8766 RTP/AVP 96 97 0 8 18 98 TECNOL O G a=rtpmap:96 eg711u/8000/1 a=rtpmap:97 eg711a/8000/1 a=rtpmap:0 pcmu/8000/1 a rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:18 g729/8000/1

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C 2011-2012 CODECs S –M as te r IC  GIPS Enhanced G.711  8kHz sampling rate D AVA N ZA D AS

 Voice Activity Detection  Variable bit rate

G ÍAS DE RE D  G.711  8kHz sampling rate  64kbps TECNOL O G _ _64kbps  G.729  8kHz sampling rate  8kHz sampling rate  8kbps

 Voice Activity Detectiony

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C 2011-2012

SIP Flows - Registration

S –M as te r IC User B _MIT.EDU_MIT.EDU Registrar MIT.EDU MIT.EDU Location D AVA N ZA D AS REGISTER: sip:[email protected] 401 - Unauthorized G ÍAS DE RE D 200 - OK

REGISTER: (add credentials)

sip:[email protected] Contact 18.18.2.4

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C 2011-2012 SIP REGISTER S –M as te r IC

REGISTER sip:mit.edu SIP/2.0

D AVA N ZA D AS p @ ; g

To: "Dennis Baron"<sip:[email protected]>;tag=324591026 Call-Id: 9ce902bd23b070ae0108b225b94ac7fa

Cseq: 5 REGISTER

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Contact: "Dennis Baron"<sip:[email protected];LINEID=05523f7a97b54dfa3f0c0e3746d73a24> Expires: 3600

Date: Thu, 30 Sep 2004 00:46:53 GMT

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Accept-Language: en

Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/2.1.11 (WinNT)g g ( )

Content-Length: 0

Via: SIP/2.0/UDP 18.10.0.79

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C 2011-2012

SIP REGISTER – 401 Response

S –M as te r IC SIP/2.0 401 Unauthorized

D AVA N ZA D AS

To: "Dennis Baron"<sip:[email protected]>;tag=324591026 Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 5 REGISTER G ÍAS DE RE D Via: SIP/2.0/UDP 18.10.0.79

Www-Authenticate: Digest realm="mit.edu", nonce="f83234924b8ae841b9b0ae8a92dcf0b71096505216", opaque="reg:change4"

D t Th 30 S 2004 00 46 56 GMT

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Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, NOTIFY, SUBSCRIBE, INFO User-Agent: Pingtel/2.2.0 (Linux)

Accept Language: en Accept-Language: en Supported: sip-cc-01, timer Content-Length: 0

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C 2011-2012

SIP REGISTER with Credentials

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REGISTER sip:mit.edu SIP/2.0

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AS To: "Dennis Baron"<sip:[email protected]>;tag=324591026 Call-Id: 9ce902bd23b070ae0108b225b94ac7fa

Cseq: 6 REGISTER

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D _{Contact: "Dennis Baron"<sip:[email protected];LINEID=05523f7a97b54dfa3f0c0e3746d73a24>}

Expires: 3600

Date: Thu, 30 Sep 2004 00:46:53 GMT

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Accept-Language: en

Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/2.1.11 (WinNT)

Content-Length: 0

Authorization: DIGEST USERNAME="[email protected]", REALM="mit.edu", NONCE="f83234924b8ae841b9b0ae8a92dcf0b71096505216", URI="sip:mit.edu", RESPONSE="ae064221a50668eaad1ff2741fa8df7d", OPAQUE="reg:change4" Via: SIP/2.0/UDP 18.10.0.79

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C 2011-2012

SIP Flows – Via Proxy

S –M as te r IC User A User B MIT.EDU MIT.EDU Proxy D AVA N ZA D AS INVITE: sip:[email protected] “Calls” dbaron @MIT.EDU INVITE: sip:[email protected] 100 - Trying G ÍAS DE RE D 180 - Ringing Rings 180 - Ringing 200 - OK Answers 200 - OK TECNOL O G ACK Talking RTP Talking BYE Hangs up g g

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C 2011-2012

SIP Flows – Via Gateway

S –M as te r IC User A _MIT.EDU_MIT.EDU Proxy 30161 Gateway D AVA N ZA D AS INVITE: sip:[email protected] “Calls” joe @MIT.EDU INVITE: sip:[email protected] 100 - Trying _Rings G ÍAS DE RE D 180 - Ringing 180 - Ringing 200 - OK 200 - OK Answers TECNOL O G ACK ACK Talking RTP Talking BYE Hangs up BYE g g 200 - OK 200 - OK 2 2

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C 2011-2012

SIP INVITE with Record-Route

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INVITE sip:[email protected] SIP/2.0

Record-Route: <sip:18.7.21.118:5080;lr;a;t=2c41;s=b07e28aa8f94660e8545313a44b9ed50> D AVA N ZA D AS

From: \"Dennis Baron\"<sip:[email protected]>;tag=2c41 To: sip:[email protected] Call-Id: [email protected] G ÍAS DE RE D Cseq: 1 INVITE

Contact: \"Dennis Baron\"<sip:[email protected]> Content-Type: application/sdp TECNOL O G Content-Length: 304 Accept-Language: en

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Supported: sip-cc, sip-cc-01, timer, replaces

User-Agent: Pingtel/2.1.11 (WinNT) Date: Thu, 30 Sep 2004 00:44:30 GMT

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C 2011-2012 SIP Standards S –M as te r IC

Just a sampling of IETF standards work…

IETF RFCshttp://ietf.org/rfc.html D AVA N ZA D AS IETF RFCshttp://ietf.org/rfc.html

 RFC3261 Core SIP specification – obsoletes RFC2543

 RFC2327 SDP – Session Description Protocol

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 RFC1889 RTP - Real-time Transport Protocol

 RFC2326 RTSP - Real-Time Streaming Protocol

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 RFC3262 SIP PRACK method – reliability for 1XX

messagesessages

 RFC3263 Locating SIP servers – SRV and NAPTR

 RFC3264 Offer/answer model for SDP use with SIP

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C 2011-2012

SIP Standards (cont.)

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 RFC3265 SIP event notification – SUBSCRIBE and

NOTIFY D AVA N ZA D AS  RFC3266 IPv6 support in SDP

 RFC3311 SIP UPDATE method – eg. changing media

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 RFC3325 Asserted identity in trusted networks

 RFC3361 Locating outbound SIP proxy with DHCP

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 RFC3428 SIP extensions for Instant Messaging

 RFC3515 SIP REFER method – eg. call transferC35 5 S et od eg ca t a s e

 SIMPLE IM/Presence

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C 2011-2012

NATs: Hole Punching - Peers tras distinto NAT

S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G 2 6

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C 2011-2012 Elements of an H.323 System S –M as te r IC  Terminals

 Multipoint Control Units (MCUs)

Referred to as “endpoints” D AVA N ZA D

AS  Multipoint Control Units (MCUs)

 Gateways  Gatekeeper G ÍAS DE RE D  Gatekeeper  Border Elements TECNOL O G

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C 2011-2012 Terminals S –M as te r IC  Telephones  Video phones D AVA N ZA D AS  Video phones  IVR devices  Voicemail Systems G ÍAS DE RE D  Voicemail Systems

 “Soft phones” (e.g., NetMeeting®)

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C 2011-2012 MCUs S –M as te r IC

 Responsible for managing multipoint conferences (two

or more endpoints engaged in a conference)

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 The MCU contains a Multipoint Controller (MC) that

manages the call signaling and may optionally have

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Multipoint Processors (MPs) to handle media mixing, switching, or other media processing

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C 2011-2012 Gateways S –M as te r IC

 The Gateway is composed of a “Media Gateway

Controller” (MGC) and a “Media Gateway” (MG), which

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AS _{may co-exist or exist separately}

 The MGC handles call signaling and other

non-media-G

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related functions

 The MG handles the media

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 Gateways interface H.323 to other networks, including

the PSTN, H.320 systems, and other H.323 networks (proxy)

(proxy)

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C 2011-2012 Gatekeeper S –M as te r IC

 The Gatekeeper is an optional component in the H.323

system which is primarily used for admission control and

D AVA N ZA D AS _{address resolution}

 The gatekeeper may allow calls to be placed directly

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between endpoints or it may route the call signaling through itself to perform functions such as follow-me/find me and forward on busy

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C 2011-2012

Border Elements and Peer Elements

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 Peer Elements, which are often co-located with a Gatekeeper,

exchange addressing information and participate in call authorization within and between administrative domains

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AS authorization within and between administrative domains

 Peer Elements may aggregate address information to reduce the

volume of routing information passed through the network

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 Border Elements are a special type of Peer Element that exists

between two administrative domains

 Border Elements may assist in call authorization/authentication

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directly between two administrative domains or via a clearinghouse

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C 2011-2012

The Protocols (cont)

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 H.323 is a “framework” document that describes how

the various pieces fit together

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 H.225.0 defines the call signaling between endpoints

and the Gatekeeper

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 RTP/RTCP (RFC 3550) is used to transmit media such as

audio and video over IP networks

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 H.225.0 Annex G and H.501 define the procedures and

protocol for communication within and between Peer Elements

Elements

 H.245 is the protocol used to control establishment and

closure of media channels within the context of a call closure of media channels within the context of a call

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C 2011-2012

The Protocols (cont)

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 H.450.x is a series of supplementary service protocols

 H.460.x is a series of version-independent extensions to

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the base H.323 protocol

 T.120 specifies how to do data conferencing

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 T.38 defines how to relay fax signals

 V.150.1 defines how to relay modem signals

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 H.235 defines security within H.323 systems

 X.680 defines the ASN.1 syntax used by the 680 de es t e S sy ta used by t e

Recommendations

 X.691 defines the Packed Encoding Rules (PER) used to g ( )

encode messages for transmission on the network 3

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C 2011-2012

Registration, Admission, and Status - RAS

S –M as te r IC  Defined in H.225.0

 Allows an endpoint to request authorization to place or

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AS  Allows an endpoint to request authorization to place or

accept a call

 Allows a Gatekeeper to control access to and from

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devices under its control

 Allows a Gatekeeper to communicate the address of

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other endpoints

 Allows two Gatekeepers to easily exchange addressing

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C 2011-2012

Registration, Admission, and Status – RAS (cont)

S –M as te r IC D AVA N ZA D AS T RRQ GK G ÍAS DE RE D _T Q _GK RCF (endpoint is registered) TECNOL O G ARQ ( p g ) ACF ACF

(endpoint may place call)

DRQ DRQ DCF

(call has terminated) T Terminal GK Gatekeeper

Symbol Key:

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C 2011-2012 The H323 stack S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G

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C 2011-2012 H323 Clients S –M as te r IC D AVA N ZA D AS

O.S. Client Price

Wi d N tM ti / f

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Unix (Linux) DC-Share nv

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Sun Sunforum +/- free

... ... ... ... ...

… ... ... ... ... ...

Y fi d bi li You can find a bigger list at:

http://www.openh323.org/h323_clients.html

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2-V

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2-V

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IP (V IP)

Voz sobre IP (VoIP)

SIP y H.323: Establecimiento y

gestión de sesiones multimedia Asterisk

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C 2011-2012 ASTERISK S –M as te r IC

 Aplicación de software libre que implementa los servicios

de una centralita telefónica de VoIP.

D AVA N ZA D AS

 Permite conectar teléfonos de VoIP (que también

pueden ser programas de ordenador o “softphones”),

í í ó ó

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fax, líneas RDSI, líneas telefónicas analógicas convencionales…

 I i i l t d ll d Li t l t

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 Inicialmente desarrollada para Linux pero actualmente

existen versiones para casi todas las plataformas.

 trixbox (con “t” minúscula) es una distribución Linux

 trixbox (con t minúscula) es una distribución Linux

(en concreto de CentOS) que incluye Asterisk y FreePBX que es un entorno gráfico basado en WEB para una

que es un entorno gráfico basado en WEB para una configuración cómoda y más sencilla de Asterisk.

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C 2011-2012 ASTERISK S –M as te r IC

 Soporta SIP, H.323, MGCP, IAX

 Se obtiene de : ftp://ftp.digium.com D AVA N ZA D AS  Se obtiene de : ftp://ftp.digium.com

 Integra casi todos los codecs de audio

 Soporte de Telefonía Tradicional

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 Soporte de Telefonía por Voz IP

 APIs para desarrollo de nuevos servicios y aplicaciones

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 Integración con Bases de Datos

 Integración con Aplicaciones ya desarrolladas

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C 2011-2012

IAX (Inter-Asterisk eXchange)

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 Actualmente en la versión 2 (IAX2) es un protocolo que

aborda el problema de los NATs.

D AVA N ZA D AS

 Utilizar el mismo puerto UDP para la señalización y para

la transmisión de los datos (RTP).

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 Simplifica el número de “agujeros” (hole-punching) a

realizar en el NAT para que el interlocutor en la intranet

l bl d d I t t

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sea alcanzable desde Internet.

 Algunos autores abogan porque IAX será el futuro de

VoIP y otros plantean que la regulación en tema de VoIP y otros plantean que la regulación en tema de

NATs, o incluso su desaparición con la entrada de IPv6 dejaran a SIP en su posición de liderato.

dejaran a SIP en su posición de liderato.

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C 2011-2012 Configuración básica S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G

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C 2011-2012 Configuración básica (2) S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G 4 4

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C 2011-2012 Configuración básica (3) S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G

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C 2011-2012 S –M as te r IC

IMPLEMENTACIÓN DE TELEFONÍA IP

Ó

D AVA N ZA D AS

EN UNA ORGANIZACIÓN

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INTEGRACIÓN CISCO-ASTERISK

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C 2011-2012

CARACTERISTICAS CISCO CALL MANAGER

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 Solución de Telefonía IP de Cisco

 Distribuible  E l bl (30000 li / id ) D AVA N ZA D AS  Escalable (30000 lineas/servidor)

 Soporta muchos usuarios

 Sobre Windows o linux

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 Soporta gran variedad de teléfonos

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C 2011-2012 PROTOCOLOS S –M as te r IC  Sip  H323 D AVA N ZA D AS  H323  MGCP (Megaco Protocol) G ÍAS DE RE D TECNOL O G 4 8

(49)

C 2011-2012 OBJETIVO FINAL S –M as te r IC 1ABC23DEF 4JKL56MNO GHI 7T UV8WXYZ9 PQRS *O PER0# 7960 CISCO IP P HON E i

messag esd irecto ries sett ing s se rvice s 12ABCDEF3 45JKLMNO6 GH I 78TUVWXYZ9 PQRS *OPER0 # 7960 CISCO IP P HONE i

message sd irecto ries setting s service s D AVA N ZA D AS 1ABC23DEF 4JKL5MNO6 GHI 7T UV8WXYZ9 PQRS *O PER0# 7960 CISCO IP P HON E i

messag esd irecto ries sett ing s se rvice s 1ABC23DEF 4 5JKL6MNO GHI 7T UV89WXYZ PQRS *O PER0# 7960 CISCO IP P HON E i

messag esd irecto ries sett ing s se rvice s G ÍAS DE RE D TECNOL O G

(50)

C 2011-2012

FUNCIONAMIENTO DE CALL MANAGER

S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G 5 0

(51)

C 2011-2012 CONFIGURACIÓN CM S –M as te r IC  Interfaz Web  https://xxxxxx/CCMAdmin/Main.asp D AVA N ZA D AS  https://xxxxxx/CCMAdmin/Main.asp G ÍAS DE RE D TECNOL O G

(52)

C 2011-2012 PARTITIONS S –M as te r IC

 Dividen el conjunto de route patterns en subconjuntos

de destinos alcanzables identificados por un nombre.

D AVA N ZA D AS

 Una partición contiene una lista de Route Patterns

 Facilitan el enrutado de llamadas dividiendo el route

G

ÍAS DE RE

D

plan en subconjuntos lógicos que se pueden basar en la organización, localización y tipo de llamada

TECNOL

O

G

5 2

(53)

C 2011-2012 Partitions S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G

(54)

C 2011-2012 SEARCH SPACES S –M as te r IC

 Es una lista ordenada de rutas de partición. Estas rutas se asocian a

los dispositivos (teléfonos).

l l d h D AVA N ZA D

AS  Determinan las particiones que los dispositivos que hacen una

llamada buscan para que esta llamada se realice

G ÍAS DE RE D TECNOL O G 5 4

(55)

C 2011-2012 ROUTE PATTERNS S –M as te r IC

 String de digitos y un conjunto de acciones

 La llamada al destino se hace solo si se marca la

D

AVA

N

ZA

D

AS  La llamada al destino se hace solo si se marca la

secuencia correcta definida en el route pattern

 Se pueden usan caracteres especiales (x…) para hacer

G

ÍAS DE RE

D p p ( ) p

rangos, etc

 Definir route patterns para diferentes tipos de llamadas:

TECNOL

O

G

(56)

C 2011-2012 ESQUEMA DE NUMERACIÓN S –M as te r IC  67xxx: Teléfonos IP HW (Vera)  68xxx: SoftPhones D AVA N ZA D AS  68xxx: SoftPhones  69xxx: Teléfonos SIP

 7xxxx: Teléfonos analógicos (fuera del Call Manager)

G

ÍAS DE RE

D  7xxxx: Teléfonos analógicos (fuera del Call Manager)

 11xxx: Teléfonos móviles TECNOL O G 5 6

(57)

C 2011-2012 Route patterns S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G

(58)

C 2011-2012 GATEWAYS S –M as te r IC

 Debe haber uno por cada campus

 Otro que será el router de salida general.

D

AVA

N

ZA

D

AS  Otro que será el router de salida general.

 Coste: 3500-4000 euros G ÍAS DE RE D TECNOL O G 5 8

(59)

C 2011-2012 Gateways S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G

(60)

C 2011-2012

TRUNK CON ASTERISK

S –M as te r IC  Es un enlace desde el Call Manager D AVA N ZA D AS el Call Manager al Asterisk:

se enrutan llamadas 14GHI2A BC5JKL3DEF6MNO

78TUV9WXYZ PQRS *0O PER# 79 6 0 CISCO IP PHO NE i

mess agesdirec tories set tings services 1A BC2DEF3 45JKLMNO6 GHI 7TUV8WXYZ9 PQRS *O PER0# 79 6 0 CISCO IP PHO NE i

mess agesdirec tories set tings services G ÍAS DE RE D se enrutan llamadas de uno al otro  Se define mediante 12A BCDEF3 45JKLMNO6 GHI 78TUVWXYZ9 PQRS *O PER0# 79 6 0 CISCO IP PHO NE i

mess agesdirec tories set tings services 12A BC3DEF 45JKL6MNO GHI 78TUV9WXYZ PQRS *0O PER# 79 6 0 CISCO IP PHO NE i

mess agesdirec tories set tings services TECNOL O G  Se define mediante la IP del Asterisk 6 0 12AB CDEF3 45JKLMNO6 GHI 78TUVWXYZ9 PQRS *0OPER# 7 960 CISCO IP PHONE i

mess agesdirectories settings s ervices 12ABCDEF3 45JKLMN O6 GHI 78TUVWXYZ9 PQRS *OPER0# 796 0 C ISCO IP PHONE i messagesdirectories sett ings services

(61)

C 2011-2012 Trunk S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G

(62)

C 2011-2012 TELEFONOS S –M as te r IC

 un identificador, el Device Name (3 caracteres más la

dirección MAC ) D AVA N ZA D AS

 una descripción (ej . la persona asociada)

 el pool al que corresponde

G

ÍAS DE RE

D p q p

 su estado (registrado o no)

 la dirección IP del teléfono: sólo se muestra si el

TECNOL

O

G

teléfono está registrado

6 2

(63)

C 2011-2012 Teléfonos S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G

(64)

C 2011-2012 Teléfonos II S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G 6 4

(65)

C 2011-2012 Teléfonos III S –M as te r IC D AVA N ZA D AS G ÍAS DE RE D TECNOL O G

Teléfono Cisco Teléfono SIP

Teléfono Cisco Teléfono SIP 300 Euros 45-50 Euros

Configuración desde el CM http://x y z w:9999/

(66)

C 2011-2012 Teléfonos IV S –M as te r IC  [69001] <--- Extensión

 username=69001 <--- Podría ser el login

 type=friend  record_out=Adhoc d dh D AVA N ZA D AS  record_in=Adhoc  qualify=no  port=5060  nat=never

 mailbox=666@testmail <--- Su buzón de voz asociado (en el voicemail.conf)

G ÍAS DE RE D @ ( )  host=dynamic  dtmfmode=info  context=from-internal  canreinvite=no  callerid=device <69001> TECNOL O G _ _{callerid=device <69001>}  language=es 6 6

(67)

C 2011-2012 Teléfonos V S –M as te r IC D AVA N ZA D AS Softphone Cisco IP C i t G ÍAS DE RE D _{IP Communicator} TECNOL O G