CHANNEL()

Sinopse:

Gets/sets various pieces of information about the canal.

Descrição:
Gets/sets various pieces of information about the canal, additional item may be available from the canal driver; see its documentation for details. Any item requested that is not available on the current canal will return an empty string. Example: Standard canal item examples; Push a hangup handler subroutine existing at dialplan ; location default,s,1 onto the current canal same => n,Set(canal(hangup_handler_push)=default,s,1) ; Set the current tonezone to Germany (de) same => n,Set(canal(tonezone)=de) ; Set the allowed maximum number of forwarding attempts same => n,Set(canal(max_forwards)=10) ; If this canal is ejected from its next bridge, and if ; the canal is not hung up, begin executing dialplan at ; location default,after-bridge,1 same => n,Set(canal(after_bridge_goto)=default,after-bridge,1) ; Log the current state of the canal same => n,Log(NOTICE, This canal is: ${canal(state)}) CALLINGPRES - Caller ID presentation for incoming calls (PRI canais) CALLINGANI2 - Caller ANI2 (PRI canais) CALLINGTON - Caller Type of Number (PRI canais) CALLINGTNS - Transit Network Selector (PRI canais) EXTEN - Current extension CONTEXT - Current context PRIORITY - Current priority canal - Current canal name UNIQUEID - Current call unique identifier HANGUPCAUSE - Asterisk cause of hangup (inbound/outbound) Technology: PJSIP Example: PJSIP specific canal examples; Log the current Call-ID same => n,Log(NOTICE, ${canal(pjsip,call-id)}) ; Log the destination address of the audio stream same => n,Log(NOTICE, ${canal(rtp,dest)}) ; Store the round-trip time associated with a ; video stream in the CDR field video-rtt same => n,Set(CDR(video-rtt)=${canal(rtcp,rtt,video)}) canal(item) item - Standard items (provided by all canal technologies) are: amaflags - R/W the Automatic Message Accounting (AMA) flags on the canal. When read from a canal, the integer value will always be returned. When written to a canal, both the string format or integer value is accepted. 1 - OMIT 2 - BILLING 3 - DOCUMENTATION accountcode - R/W the canal's account code. audioreadformat - R/O format currently being read. audionativeformat - R/O format used natively for audio. audiowriteformat - R/O format currently being written. dtmf_features - R/W The canal's DTMF bridge features. May include one or more of 'T' 'K' 'H' 'W' and 'X' in a similar manner to options in the 'Dial' aplicação. When setting it, the features string must be all upper case. callgroup - R/W numeric call pickup groups that this canal is a member. pickupgroup - R/W numeric call pickup groups this canal can pickup. namedcallgroup - R/W named call pickup groups that this canal is a member. namedpickupgroup - R/W named call pickup groups this canal can pickup. channeltype - R/O technology used for canal. checkhangup - R/O Whether the canal is hanging up (1/0) digitdetect - R/W Enable or disable DTMF detection on canal drivers that support it. If set on a DAHDI canal, this will only disable DTMF detection, not pulse dialing detection. To disable pulse dialing, use the 'dialmode' option. On DAHDI canais, this will disable DSP if it is not needed for anything else. This will prevent DTMF detection regardless of the 'dialmode' setting. faxdetect - R/W Enable or disable fax detection on canal drivers that support it. after_bridge_goto - R/W the parseable goto string indicating where the canal is expected to return to in the PBX after exiting the next bridge it joins on the condition that it doesn't hang up. The parseable goto string uses the same syntax as the 'Goto' aplicação. hangup_handler_pop - W/O Replace the most recently added hangup handler with a new hangup handler on the canal if supplied. The assigned string is passed to the Gosub aplicação when the canal is hung up. Any optionally omitted context and exten are supplied by the canal pushing the handler before it is pushed. hangup_handler_push - W/O Push a hangup handler onto the canal hangup handler stack. The assigned string is passed to the Gosub aplicação when the canal is hung up. Any optionally omitted context and exten are supplied by the canal pushing the handler before it is pushed. hangup_handler_wipe - W/O Wipe the entire hangup handler stack and replace with a new hangup handler on the canal if supplied. The assigned string is passed to the Gosub aplicação when the canal is hung up. Any optionally omitted context and exten are supplied by the canal pushing the handler before it is pushed. onhold - R/O Whether or not the canal is onhold. (1/0) language - R/W language for sounds played. musicclass - R/W class (from musiconhold.conf) for hold music. name - The name of the canal parkinglot - R/W parkinglot for parking. relaxdtmf - W/O Enable or disable relaxed DTMF detection for canal drivers that support it, overriding any setting previously defaulted by the canal driver. rxgain - R/W set rxgain level on canal drivers that support it. secure_bridge_signaling - Whether or not canais bridged to this canal require secure signaling (1/0) secure_bridge_media - Whether or not canais bridged to this canal require secure media (1/0) state - R/O state of the canal tdd - R/W Enable or disable TDD mode on canal drivers that support it. When reading this option, 1 indicates TDD mode enabled, 0 indicates TDD mode disabled, and 'mate' indicates TDD mate mode. tonezone - R/W zone for indications played transfercapability - R/W ISDN Transfer Capability, one of: SPEECH DIGITAL RESTRICTED_DIGITAL 3K1AUDIO DIGITAL_W_TONES VIDEO txgain - R/W set txgain level on canal drivers that support it. videonativeformat - R/O format used natively for video hangupsource - R/W returns the canal responsible for hangup. appname - R/O returns the internal aplicação name. appdata - R/O returns the aplicação data if available. exten - R/O returns the extension for an outbound canal. context - R/O returns the context for an outbound canal. lastexten - R/O returns the last unique extension for an outbound canal. lastcontext - R/O returns the last unique context for an outbound canal. channame - R/O returns the canal name for an outbound canal. uniqueid - R/O returns the canal uniqueid. linkedid - R/O returns the linkedid if available, otherwise returns the uniqueid. tenantid - R/W The canal tenantid. max_forwards - R/W The maximum number of forwards allowed. callid - R/O Call identifier log tag associated with the canal e.g., '[C-00000000]'. Technology: DAHDI dahdi_channel - R/O DAHDI canal related to this canal. dahdi_span - R/O DAHDI span related to this canal. dahdi_group - R/O DAHDI logical group related to this canal. dahdi_type - R/O DAHDI canal type, one of: analog mfc/r2 pri pseudo ss7 keypad_digits - R/O PRI Keypad digits that came in with the SETUP message. reversecharge - R/O PRI Reverse Charging Indication, one of: -1 - None 1 - Reverse Charging Requested no_media_path - R/O PRI Nonzero if the canal has no B canal. The canal is either on hold or a call waiting call. buffers - W/O Change the canal's buffer policy (for the current call only) This option takes two arguments: Number of buffers, Buffer policy being one of: 'full' 'immediate' 'half' echocan_mode - W/O Change the configuration of the active echo canceller on the canal (if any), for the current call only. Possible values are: 'on'Normal mode (the echo canceller is actually reinitialized) 'off'Disabled 'fax'FAX/data mode (NLP disabled if possible, otherwise completely disabled) 'voice'Voice mode (returns from FAX mode, reverting the changes that were made) dialmode - R/W Pulse and tone dialing mode of the canal. Disabling tone dialing using this option will not disable the DSP used for DTMF detection. To do that, also set the 'digitdetect' option. If digit detection is disabled, DTMF will not be detected, regardless of the 'dialmode' setting. The 'digitdetect' setting has no impact on pulse dialing detection. If set, overrides the setting in 'chan_dahdi.conf' for that canal. both pulse dtmf tone none waitfordialtone - W/O Duration in ms for which to wait for dial tone on the current call. This setting is will temporarily override the 'waitfordialtone' setting in 'chan_dahdi.conf' (typically if that setting is disabled). You must call this in a pre-dial handler when making a call on an analog trunk (e.g. FXS-signalled interface). This allows, for example, being able to barge in on an in-use trunk, if dialed specifically, but allows skipping the trunk when routing calls if dial tone is not present on a canal. This setting will only apply to the current (next) call made on the DAHDI canal, and will not persist for future calls. Please keep in mind that due to the way that chan_dahdi implements dial tone detection, DTMF digits on an in-use canal will temporarily relay to any other canais attempting to use the canal for a call. However, voice transmission will not leak. Technology: IAX osptoken - R/O Get the peer's osptoken. peerip - R/O Get the peer's ip address. peername - R/O Get the peer's username. secure_signaling - R/O Get the if the IAX canal is secured. secure_media - R/O Get the if the IAX canal is secured. Technology: OOH323 faxdetect - R/W Fax Detect Returns 0 or 1 Write yes or no t38support - R/W t38support Returns 0 or 1 Write yes or no h323id_url - R/O Returns caller URL caller_h323id - R/O Returns caller h323id caller_dialeddigits - R/O Returns caller dialed digits caller_email - R/O Returns caller email callee_email - R/O Returns callee email callee_dialeddigits - R/O Returns callee dialed digits caller_url - R/O Returns caller URL max_forwards - R/W Get or set the maximum number of call forwards for this canal. This number describes the number of times a call may be forwarded by this canal before the call fails. "Forwards" in this case refers to redirects by phones as well as calls to local canais. Note that this has no relation to the SIP Max-Forwards header. Technology: PJSIP rtp - R/O Retrieve media related information. type - When rtp is specified, the 'type' parameter must be provided. It specifies which RTP parameter to read. src - Retrieve the local address for RTP. dest - Retrieve the remote address for RTP. direct - If direct media is enabled, this address is the remote address used for RTP. secure - Whether or not the media stream is encrypted. 0 - The media stream is not encrypted. 1 - The media stream is encrypted. hold - Whether or not the media stream is currently restricted due to a call hold. 0 - The media stream is not held. 1 - The media stream is held. media_type - When rtp is specified, the 'media_type' parameter may be provided. It specifies which media stream the chosen RTP parameter should be retrieved from. audio - Retrieve information from the audio media stream. Note If not specified, 'audio' is used by default. video - Retrieve information from the video media stream. rtcp - R/O Retrieve RTCP statistics. statistic - When rtcp is specified, the 'statistic' parameter must be provided. It specifies which RTCP statistic parameter to read. all - Retrieve a summary of all RTCP statistics. The following data items are returned in a semi-colon delineated list: ssrc - Our Synchronization Source identifier themssrc - Their Synchronization Source identifier lp - Our lost packet count rxjitter - Received packet jitter rxcount - Received packet count txjitter - Transmitted packet jitter txcount - Transmitted packet count rlp - Remote lost packet count rtt - Round trip time txmes - Transmitted Media Experience Score rxmes - Received Media Experience Score all_jitter - Retrieve a summary of all RTCP Jitter statistics. The following data items are returned in a semi-colon delineated list: minrxjitter - Our minimum jitter maxrxjitter - Our max jitter avgrxjitter - Our average jitter stdevrxjitter - Our jitter standard deviation reported_minjitter - Their minimum jitter reported_maxjitter - Their max jitter reported_avgjitter - Their average jitter reported_stdevjitter - Their jitter standard deviation all_loss - Retrieve a summary of all RTCP packet loss statistics. The following data items are returned in a semi-colon delineated list: minrxlost - Our minimum lost packets maxrxlost - Our max lost packets avgrxlost - Our average lost packets stdevrxlost - Our lost packets standard deviation reported_minlost - Their minimum lost packets reported_maxlost - Their max lost packets reported_avglost - Their average lost packets reported_stdevlost - Their lost packets standard deviation all_rtt - Retrieve a summary of all RTCP round trip time information. The following data items are returned in a semi-colon delineated list: minrtt - Minimum round trip time maxrtt - Maximum round trip time avgrtt - Average round trip time stdevrtt - Standard deviation round trip time all_mes - Retrieve a summary of all RTCP Media Experience Score information. The following data items are returned in a semi-colon delineated list: minmes - Minimum MES based on us analysing received packets. maxmes - Maximum MES based on us analysing received packets. avgmes - Average MES based on us analysing received packets. stdevmes - Standard deviation MES based on us analysing received packets. reported_minmes - Minimum MES based on data we get in Sender and Receiver Reports sent by the remote end reported_maxmes - Maximum MES based on data we get in Sender and Receiver Reports sent by the remote end reported_avgmes - Average MES based on data we get in Sender and Receiver Reports sent by the remote end reported_stdevmes - Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end txcount - Transmitted packet count rxcount - Received packet count txjitter - Transmitted packet jitter rxjitter - Received packet jitter remote_maxjitter - Their max jitter remote_minjitter - Their minimum jitter remote_normdevjitter - Their average jitter remote_stdevjitter - Their jitter standard deviation local_maxjitter - Our max jitter local_minjitter - Our minimum jitter local_normdevjitter - Our average jitter local_stdevjitter - Our jitter standard deviation txploss - Transmitted packet loss rxploss - Received packet loss remote_maxrxploss - Their max lost packets remote_minrxploss - Their minimum lost packets remote_normdevrxploss - Their average lost packets remote_stdevrxploss - Their lost packets standard deviation local_maxrxploss - Our max lost packets local_minrxploss - Our minimum lost packets local_normdevrxploss - Our average lost packets local_stdevrxploss - Our lost packets standard deviation rtt - Round trip time maxrtt - Maximum round trip time minrtt - Minimum round trip time normdevrtt - Average round trip time stdevrtt - Standard deviation round trip time local_ssrc - Our Synchronization Source identifier remote_ssrc - Their Synchronization Source identifier txmes - Current MES based on us analyzing rtt, jitter and loss in the actual received RTP stream received from the remote end. I.E. This is the MES for the incoming audio stream. rxmes - Current MES based on rtt and the jitter and loss values in RTCP sender and receiver reports we receive from the remote end. I.E. This is the MES for the outgoing audio stream. remote_maxmes - Max MES based on data we get in Sender and Receiver Reports sent by the remote end remote_minmes - Min MES based on data we get in Sender and Receiver Reports sent by the remote end remote_normdevmes - Average MES based on data we get in Sender and Receiver Reports sent by the remote end remote_stdevmes - Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end local_maxmes - Max MES based on us analyzing the received RTP stream local_minmes - Min MES based on us analyzing the received RTP stream local_normdevmes - Average MES based on us analyzing the received RTP stream local_stdevmes - Standard deviation MES based on us analyzing the received RTP stream media_type - When rtcp is specified, the 'media_type' parameter may be provided. It specifies which media stream the chosen RTCP parameter should be retrieved from. audio - Retrieve information from the audio media stream. Note If not specified, 'audio' is used by default. video - Retrieve information from the video media stream. endpoint - R/O The name of the endpoint associated with this canal. Use the PJSIP_ENDPOINT função to obtain further endpoint related information. contact - R/O The name of the contact associated with this canal. Use the PJSIP_CONTACT função to obtain further contact related information. Note this may not be present and if so is only available on outgoing legs. aor - R/O The name of the AOR associated with this canal. Use the PJSIP_AOR função to obtain further AOR related information. Note this may not be present and if so is only available on outgoing legs. pjsip - R/O Obtain information about the current PJSIP canal and its session. type - When pjsip is specified, the 'type' parameter must be provided. It specifies which signalling parameter to read. call-id - The SIP call-id. secure - Whether or not the signalling uses a secure transport. 0 - The signalling uses a non-secure transport. 1 - The signalling uses a secure transport. target_uri - The contact URI where requests are sent. local_uri - The local URI. local_tag - Tag in From header remote_uri - The remote URI. remote_tag - Tag in To header request_uri - The request URI of the incoming 'INVITE' associated with the creation of this canal. t38state - The current state of any T.38 fax on this canal. DISABLED - T.38 faxing is disabled on this canal. LOCAL_REINVITE - Asterisk has sent a 're-INVITE' to the remote end to initiate a T.38 fax. REMOTE_REINVITE - The remote end has sent a 're-INVITE' to Asterisk to initiate a T.38 fax. ENABLED - A T.38 fax session has been enabled. REJECTED - A T.38 fax session was attempted but was rejected. local_addr - On inbound calls, the full IP address and port number that the 'INVITE' request was received on. On outbound calls, the full IP address and port number that the 'INVITE' request was transmitted from. remote_addr - On inbound calls, the full IP address and port number that the 'INVITE' request was received from. On outbound calls, the full IP address and port number that the 'INVITE' request was transmitted to. Technology: SIP peerip - R/O Get the IP address of the peer. recvip - R/O Get the source IP address of the peer. recvport - R/O Get the source port of the peer. from - R/O Get the URI from the From: header. uri - R/O Get the URI from the Contact: header. ruri - R/O Get the Request-URI from the INVITE header. useragent - R/O Get the useragent. peername - R/O Get the name of the peer. t38passthrough - R/O '1' if T38 is offered or enabled in this canal, otherwise '0' rtpqos - R/O Get QOS information about the RTP stream This option takes two additional arguments: Argument 1: 'audio' Get data about the audio stream 'video' Get data about the video stream 'text' Get data about the text stream Argument 2: 'local_ssrc' Local SSRC (stream ID) 'local_lostpackets' Local lost packets 'local_jitter' Local calculated jitter 'local_maxjitter' Local calculated jitter (maximum) 'local_minjitter' Local calculated jitter (minimum) 'local_normdevjitter'Local calculated jitter (normal deviation) 'local_stdevjitter' Local calculated jitter (standard deviation) 'local_count' Number of received packets 'remote_ssrc' Remote SSRC (stream ID) 'remote_lostpackets'Remote lost packets 'remote_jitter' Remote reported jitter 'remote_maxjitter' Remote calculated jitter (maximum) 'remote_minjitter' Remote calculated jitter (minimum) 'remote_normdevjitter'Remote calculated jitter (normal deviation) 'remote_stdevjitter'Remote calculated jitter (standard deviation) 'remote_count' Number of transmitted packets 'rtt' Round trip time 'maxrtt' Round trip time (maximum) 'minrtt' Round trip time (minimum) 'normdevrtt' Round trip time (normal deviation) 'stdevrtt' Round trip time (standard deviation) 'all' All statistics (in a form suited to logging, but not for parsing) rtpdest - R/O Get remote RTP destination information. This option takes one additional argument: Argument 1: 'audio' Get audio destination 'video' Get video destination 'text' Get text destination Defaults to 'audio' if unspecified. rtpsource - R/O Get source RTP destination information. This option takes one additional argument: Argument 1: 'audio' Get audio destination 'video' Get video destination 'text' Get text destination Defaults to 'audio' if unspecified. This documentation was generated from Asterisk branch 18 using version GIT

Sintaxe:

; Push a hangup handler subroutine existing at dialplan

; location default,s,1 onto the current channel

same => n,Set(CHANNEL(hangup_handler_push)=default,s,1)

; Set the current tonezone to Germany (de)

same => n,Set(CHANNEL(tonezone)=de)

; Set the allowed maximum number of forwarding attempts

same => n,Set(CHANNEL(max_forwards)=10)

; If this channel is ejected from its next bridge, and if

; the channel is not hung up, begin executing dialplan at

; location default,after-bridge,1

same => n,Set(CHANNEL(after_bridge_goto)=default,after-bridge,1)

; Log the current state of the channel

same => n,Log(NOTICE, This channel is: ${CHANNEL(state)})

; Log the current Call-ID

same => n,Log(NOTICE, ${CHANNEL(pjsip,call-id)})

; Log the destination address of the audio stream

same => n,Log(NOTICE, ${CHANNEL(rtp,dest)})

; Store the round-trip time associated with a

; video stream in the CDR field video-rtt

same => n,Set(CDR(video-rtt)=${CHANNEL(rtcp,rtt,video)})

CHANNEL(item)

Fonte: Asterisk Wiki