View source for MGCP call agent module


This module implements the Call Agent part of [http://en.wikipedia.org/wiki/Media_Gateway_Control_Protocol_%28MGCP%29 MGCP] allowing Yate to control analog or digital Media Gateways.

The resources of these gateways appear as voice circuits so you need to reference them in the [http://docs.yate.ro/wiki/Analog analog] or [[Ysigchan|signaling module]].

Specific information is available for configuring digital voice trunks for [[Configure Mediant MGCP|Mediant]] and [[Configure Cisco MGCP|Cisco]] gateways.

===Configuration file===
 mgcpca.conf:
 
 [general]
 ; This section sets global variables of the MGCP Call Agent module
 
 ; priority: int: Priority of the chan.rtp message handler
 ;priority=80
 
 ; tzoffset: int: Timezone offset from GMT in seconds for sending Caller ID
 ; Positive to go East, negative to go West
 ; Examples: CET=3600, EST=-18000
 ; This parameter is reloadable
 ;tzoffset=0
 
 [engine]
 ; These parameters are used to initialize the MGCP Engine
 
 ; enabled: bool: Enable the MGCP engine
 ;enabled=yes
 
 ; address: ipaddress: Local IP address to bind to, default all interfaces
 ;address=0.0.0.0
 
 ; port: int: UDP port to bind, set to zero to use default MGCP CA port (2727)
 ;port=
 
 ; request_ack: bool: Request an Acknowledge of the transactions
 ;request_ack=yes
 
 ; send_provisional: bool: Send 1xx provisional answers to the requests
 ;send_provisional=yes
 
 
 [endpoint]
 ; Settings for the local endpoint
 
 ; user: string: The user part of the local endpoint
 ;user=yate
 
 ; host: string: The host part of the local endpoint, default engine's address
 ;host=
 
 ; port: int: Declared local port, defaults to engine's port
 ;port=
 
 
 [codecs]
 ; This section allows to individually enable or disable the codecs
 
 ; default: bool: Enable all unlisted codecs by default if a transcoder exists
 ;default=enable
 
 ; mulaw: bool: Companded-only G711 mu-law (PCMU/8000)
 ;mulaw=default
 
 ; alaw: bool: Companded-only G711 a-law (PCMU/8000)
 ;alaw=default
 
 ; gsm: bool: European GSM 06.10 (GSM/8000)
 ;gsm=default
 
 ; lpc10: bool: Linear Prediction Codec (LPC/8000)
 ;lpc10=default
 
 ; ilbc: bool: Internet Low Bandwidth Codec (iLBC/8000)
 ;ilbc=default
 
 ; amr: bool: Adaptive Multi-Rate 3GPP (AMR/8000)
 ;amr=default
 
 ; slin: bool: Signed Linear 16-bit uncompressed (L16/8000)
 ;slin=default
 
 ; g723: bool: ITU G.723 all variations (G723/8000)
 ;g723=default
 
 ; g726: bool: ITU G.726 32-bit (G726-32/8000)
 ;g726=default
 
 ; g728: bool: ITU G.728 all variations (G728/8000)
 ;g728=default
 
 ; g729: bool: ITU G.729 all variations (G729/8000)
 ;g729=default
 
 ; g729_annexb: bool: G.729 Annex B (VAD) support default (if not in SDP)
 ; NOTE: RFC 3555 specifies the default should be yes
 ;g729_annexb=no
 
 ; amr_octet: bool: Octet aligned AMR RTP payload default (if not in SDP)
 ; NOTE: RFC 4867 (and older 3267) specifies the default is bandwidth efficient
 ;amr_octet=no
 
 [hacks]
 ; This section holds the dirty stuff required to work with some broken
 ;  implementations
 ;
 ; ilbc_forced: string: Format to force as iLBC, can be: ilbc20 or ilbc30
 ;ilbc_forced=
 ;
 ; ilbc_default: string: Format to use for iLBC when packetization is unknown
 ;ilbc_default=ilbc30
 
 ; g729_annexb: bool: Force G.729 Annex B support when parsing the SDP
 ;g729_annexb=
 
 ; ignore_sdp_port: bool: Ignore SDP changes if only the port is different
 ; This allows preserving the local RTP session and port
 ;ignore_sdp_port=no
 
 ;[gw PUT_NAME_HERE]
 ; One gw ... section is required for each gateway we control
 ; The gateway name is a string that can be referenced from other modules
 
 ; user: string: Remote MGCP resource name
 ;user=
 
 ; host: ipaddress: IP address or hostname of the gateway, mandatory
 ;host=
 
 ; port: int: Remote port to send packets to
 ;port=2427
 
 ; address: ipaddress: IP address of the gateway, if present will override
 ;  resolving the host name and will immediately make the span operational
 ;  without waiting for a remote RSIP
 ;address=
 
 ; version: string: Protocol name and version to use initially, if left blank
 ;  will autodetect from first received RSIP, until then uses MGCP 1.0
 ; Depending on remote capabilities and settings it is possible to use the SGCP
 ;  or TGCP protocols too.
 ;version=
 
 ; range: string: Range of gateways to create using this section as a template
 ; If range is not empty the "user" setting must contain an asterisk
 ; Example for creating aaln1/[1-10], aaln2/[1-10], aaln3/[1-10] and aaln6/[1-10]:
 ;  user=aaln*/1
 ;  chans=10
 ;  range=1-3,6
 ;range=
 
 ; cluster: bool: This is the gateway to send remote RTP commands to
 ; Do not activate this functionality unless you know very well what it does
 ;  as it disrupts other based RTP protocols
 ;cluster=no
 
 ; voicechans: string: The range of channels used for voice (data) transfer
 ; Channel intervals may be specified by separating the margins using the '-' character
 ; Individual channels or channel intervals may be separated by '.'
 ; Duplicate or not strict positive channel numbers are not allowed
 ; If specified the highest circuit number is used as number of PSTN channels
 ; E.g. 1-15.17-31 or 1-23 or 1.2.3-15.17.18-30.31
 ;voicechans=
 
 ; chans: int: Number of PSTN circuits to create in span
 ; Defaults to 1 or highest voicechan declared
 ;chans=1
 
 ; offset: int: Offset of circuit code relative to start in span
 ;offset=0
 
 ; increment: integer: How much will increment this span the circuit number in
 ;  the group it belongs to
 ; Defaults to 32 if chans is 30 or 31 (E1), 24 if chans is 23 (T1), number of
 ;  circuits in all other cases
 ;increment=
 
 ; clearconn: bool: Clear all connections of remote endpoints when initialized
 ;clearconn=no
 
 ; bearer: keyword: Default bearer encoding
 ; Allowed values: mulaw, alaw
 ; If bearer is set a default encoding is assumed on the circuit side and a
 ;  B: e:mu or B: e:A line is added only if the circuit data format changes
 ; Normally the default bearer is not set so a B: line is added when needed
 ;bearer=
 
 ; req_dtmf: keyword: When to request DTMF notification from gateway
 ; Allowed values:
 ;  none - do not request to be notified about DTMF events
 ;  once - emit RQNT once when connecting the circuit (default)
 ;  more - emit RQNT on connection and after each notification received
 ;req_dtmf=once
 
 ; forward_rtp: bool: Support to forward RTP directly to protocols that support it
 ; The actual direct RTP forwarding is still negotiated in routing
 ;forward_rtp=yes for digital gateways, no for analogic
 
 ; forward_sdp: bool: Also include the raw SDP in the RTP forward offer
 ;forward_sdp=no



===Examples===

Configuration for a Mediant MP-114 FXS analog gateway.

mgcpca.conf:
 
 [general]
 ; Central European Time, +0100
 tzoffset=3600
 
 [engine]
 request_ack=no
 send_provisional=no
 
 [endpoint]
 
 [gw mp114]
 host=10.0.0.1
 address=10.0.0.1
 user=ACgw0
 chans=4
 default=no
 clearconn=yes


analog.conf:

 [mediant]
 type=FXS
 spans=mp116
 callsetup=before
 ringback=yes


regexroute.conf:

 [priorities]
 prerouteall=yes
 
 [contexts]
 ; set caller number depending on line on Mediant
 ${address}^mediant/\([0-3]\)$=;caller=18\1
 
 [default]
 ; enable RTP forwarding if supported by caller channel
 ${rtp_forward}possible=;rtp_forward=true
 ; perform overlapped dial
 ^off-hook$=external/nodata/overlapped.php;accept_call=true
 ; route 18x to the corresponding port
 ^18\([0-3]\)$=analog/mediant/\1
 ; ...
 ; If we reached this point and not using overlapped dialing return without route
 ${overlapped}yes^=return
 ; If number is too long to be possibly valid error out
 .\{20\}=-;error=noroute
 ; Let the dialer know there may be other routes
 .*=;error=incomplete

'''See also'''

* [http://docs.yate.ro/wiki/Analog Analog | Analog module]
* [[Ysigchan|Signaling module]]
* [[Configure Mediant MGCP]]
* [[Configure Cisco MGCP]]