Hangup an incoming PJSIP channel with a SIP response code May be one of... A numeric response code in the range 400 ->699 A response code name from third-party/pjproject/source/pjsip/include/pjsip/sip_msg.h such as USE_IDENTITY_HEADER or PJSIP_SC_USE_IDENTITY_HEADER Hangs up an incoming PJSIP channel and returns the specified SIP response code in the final response to the caller. This function must be called BEFORE anything that might cause any other final (non 1XX) response to be sent. For example calling Answer() or Playback without the noanswer option will cause the call to be answered and a final 200 response to be sent. As with the Hangup application, the dialplan will terminate after calling this function. The cause code set on the channel will be translated to a standard ISDN cause code using the table defined in ast_sip_hangup_sip2cause() in res_pjsip.c same = n,PJSIPHangup(437) same = n,PJSIPHangup(UNSUPPORTED_CERTIFICATE) same = n,ExecIf($[${SOMEVALUE} = ${SOME_BAD_VALUE}]?PJSIPHangup(437)) Hangup an incoming PJSIP channel with a SIP response code Hangs up an incoming PJSIP channel and returns the specified SIP response code in the final response to the caller. This function must be called BEFORE anything that might cause any other final (non 1XX) response to be sent. For example calling Answer() or Playback without the noanswer option will cause the call to be answered and a final 200 response to be sent. The cause code set on the channel will be translated to a standard ISDN cause code using the table defined in ast_sip_hangup_sip2cause() in res_pjsip.c Action: PJSIPHangup ActionID: 12345678 Channel: PJSIP/alice-00000002 Cause: 437 Action: PJSIPHangup ActionID: 12345678 Channel: PJSIP/alice-00000002 Cause: UNSUPPORTED_CERTIFICATE Return a dial string for dialing all contacts on an AOR. Name of the endpoint Name of an AOR to use, if not specified the configured AORs on the endpoint are used Optional request user to use in the request URI Returns a properly formatted dial string for dialing all contacts on an AOR. Media and codec offerings to be set on an outbound SIP channel prior to dialing. types of media offered When read, returns the codecs offered based upon the media choice. When written, sets the codecs to offer when an outbound dial attempt is made, or when a session refresh is sent using PJSIP_SEND_SESSION_REFRESH. PJSIP_SEND_SESSION_REFRESH 13.18.0 14.7.0 15.1.0 16.0.0 Get or change the DTMF mode for a SIP call. When read, returns the current DTMF mode When written, sets the current DTMF mode This function uses the same DTMF mode naming as the dtmf_mode configuration option Get or change the on-hold behavior for a SIP call. When read, returns the current moh passthrough mode When written, sets the current moh passthrough mode If yes, on-hold re-INVITEs are sent. If no, music on hold is generated. This function can be used to override the moh_passthrough configuration option 13.12.0 14.1.0 15.0.0 W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session The type of update to send. Default is invite. Send the session refresh as a re-INVITE. Send the session refresh as an UPDATE. This function will cause the PJSIP stack to immediately refresh the media session for the channel. This will be done using either a re-INVITE (default) or an UPDATE request. This is most useful when combined with the PJSIP_MEDIA_OFFER dialplan function, as it allows the formats in use on a channel to be re-negotiated after call setup. The formats the endpoint supports are not checked or enforced by this function. Using this function to offer formats not supported by the endpoint may result in a loss of media. ; Within some existing extension on an answered channel same => n,Set(PJSIP_MEDIA_OFFER(audio)=!all,g722) same => n,Set(PJSIP_SEND_SESSION_REFRESH()=invite) PJSIP_MEDIA_OFFER 13.24.0 16.1.0 17.0.0 Parse an uri and return a type part of the URI. URI to parse The type parameter specifies which URI part to read Display name. URI scheme. User part. Password part. Host part. Port number, or zero. User parameter. Method parameter. Transport parameter. TTL param, or -1. Loose routing param, or zero. Maddr param. Parse an URI and return a specified part of the URI. R/O Retrieve media related information. When rtp is specified, the type parameter must be provided. It specifies which RTP parameter to read. Retrieve the local address for RTP. Retrieve the remote address for RTP. If direct media is enabled, this address is the remote address used for RTP. Whether or not the media stream is encrypted. The media stream is not encrypted. The media stream is encrypted. Whether or not the media stream is currently restricted due to a call hold. The media stream is not held. The media stream is held. When rtp is specified, the media_type parameter may be provided. It specifies which media stream the chosen RTP parameter should be retrieved from. Retrieve information from the audio media stream. If not specified, audio is used by default. Retrieve information from the video media stream. R/O Retrieve RTCP statistics. When rtcp is specified, the statistic parameter must be provided. It specifies which RTCP statistic parameter to read. Retrieve a summary of all RTCP statistics. The following data items are returned in a semi-colon delineated list: Our Synchronization Source identifier Their Synchronization Source identifier Our lost packet count Received packet jitter Received packet count Transmitted packet jitter Transmitted packet count Remote lost packet count Round trip time Transmitted Media Experience Score Received Media Experience Score Retrieve a summary of all RTCP Jitter statistics. The following data items are returned in a semi-colon delineated list: Our minimum jitter Our max jitter Our average jitter Our jitter standard deviation Their minimum jitter Their max jitter Their average jitter Their jitter standard deviation Retrieve a summary of all RTCP packet loss statistics. The following data items are returned in a semi-colon delineated list: Our minimum lost packets Our max lost packets Our average lost packets Our lost packets standard deviation Their minimum lost packets Their max lost packets Their average lost packets Their lost packets standard deviation Retrieve a summary of all RTCP round trip time information. The following data items are returned in a semi-colon delineated list: Minimum round trip time Maximum round trip time Average round trip time Standard deviation round trip time Retrieve a summary of all RTCP Media Experience Score information. The following data items are returned in a semi-colon delineated list: Minimum MES based on us analysing received packets. Maximum MES based on us analysing received packets. Average MES based on us analysing received packets. Standard deviation MES based on us analysing received packets. Minimum MES based on data we get in Sender and Receiver Reports sent by the remote end Maximum MES based on data we get in Sender and Receiver Reports sent by the remote end Average MES based on data we get in Sender and Receiver Reports sent by the remote end Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end Transmitted packet count Received packet count Transmitted packet jitter Received packet jitter Their max jitter Their minimum jitter Their average jitter Their jitter standard deviation Our max jitter Our minimum jitter Our average jitter Our jitter standard deviation Transmitted packet loss Received packet loss Their max lost packets Their minimum lost packets Their average lost packets Their lost packets standard deviation Our max lost packets Our minimum lost packets Our average lost packets Our lost packets standard deviation Round trip time Maximum round trip time Minimum round trip time Average round trip time Standard deviation round trip time Our Synchronization Source identifier Their Synchronization Source identifier 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. 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. Max MES based on data we get in Sender and Receiver Reports sent by the remote end Min MES based on data we get in Sender and Receiver Reports sent by the remote end Average MES based on data we get in Sender and Receiver Reports sent by the remote end Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end Max MES based on us analyzing the received RTP stream Min MES based on us analyzing the received RTP stream Average MES based on us analyzing the received RTP stream Standard deviation MES based on us analyzing the received RTP stream When rtcp is specified, the media_type parameter may be provided. It specifies which media stream the chosen RTCP parameter should be retrieved from. Retrieve information from the audio media stream. If not specified, audio is used by default. Retrieve information from the video media stream. R/O The name of the endpoint associated with this channel. Use the PJSIP_ENDPOINT function to obtain further endpoint related information. R/O The name of the contact associated with this channel. Use the PJSIP_CONTACT function to obtain further contact related information. Note this may not be present and if so is only available on outgoing legs. R/O The name of the AOR associated with this channel. Use the PJSIP_AOR function to obtain further AOR related information. Note this may not be present and if so is only available on outgoing legs. R/O Obtain information about the current PJSIP channel and its session. When pjsip is specified, the type parameter must be provided. It specifies which signalling parameter to read. The SIP call-id. Whether or not the signalling uses a secure transport. The signalling uses a non-secure transport. The signalling uses a secure transport. The contact URI where requests are sent. The local URI. Tag in From header The remote URI. Tag in To header The request URI of the incoming INVITE associated with the creation of this channel. The current state of any T.38 fax on this channel. T.38 faxing is disabled on this channel. Asterisk has sent a re-INVITE to the remote end to initiate a T.38 fax. The remote end has sent a re-INVITE to Asterisk to initiate a T.38 fax. A T.38 fax session has been enabled. A T.38 fax session was attempted but was rejected. 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. 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. ; 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)})