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                            Asterisk Project : Asterisk 13 Function_PJSIP_ENDPOINT
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            Created by <span class='author'> wikibot</span> on Aug 08, 2014
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                    <h1 id="Asterisk13Function_PJSIP_ENDPOINT-PJSIP_ENDPOINT()">PJSIP_ENDPOINT()</h1>

<h3 id="Asterisk13Function_PJSIP_ENDPOINT-Synopsis">Synopsis</h3>

<p>Get information about a PJSIP endpoint</p>

<h3 id="Asterisk13Function_PJSIP_ENDPOINT-Description">Description</h3>

<h3 id="Asterisk13Function_PJSIP_ENDPOINT-Syntax">Syntax</h3>


<div class="preformatted panel" style="border-width: 1px;"><div class="preformattedContent panelContent">
<pre>PJSIP_ENDPOINT(name,field)</pre>
</div></div>

<h5 id="Asterisk13Function_PJSIP_ENDPOINT-Arguments">Arguments</h5>

<ul>
	<li><code>name</code> - The name of the endpoint to query.</li>
	<li><code>field</code> - The configuration option for the endpoint to query for. Supported options are those fields on the <em>endpoint</em> object in <code>pjsip.conf</code>.
	<ul>
		<li><code>100rel</code> - Allow support for RFC3262 provisional ACK tags</li>
		<li><code>aggregate_mwi</code> - Condense MWI notifications into a single NOTIFY.</li>
		<li><code>allow</code> - Media Codec(s) to allow</li>
		<li><code>aors</code> - AoR(s) to be used with the endpoint</li>
		<li><code>auth</code> - Authentication Object(s) associated with the endpoint</li>
		<li><code>callerid</code> - CallerID information for the endpoint</li>
		<li><code>callerid_privacy</code> - Default privacy level</li>
		<li><code>callerid_tag</code> - Internal id_tag for the endpoint</li>
		<li><code>context</code> - Dialplan context for inbound sessions</li>
		<li><code>direct_media_glare_mitigation</code> - Mitigation of direct media (re)INVITE glare</li>
		<li><code>direct_media_method</code> - Direct Media method type</li>
		<li><code>connected_line_method</code> - Connected line method type</li>
		<li><code>direct_media</code> - Determines whether media may flow directly between endpoints.</li>
		<li><code>disable_direct_media_on_nat</code> - Disable direct media session refreshes when NAT obstructs the media session</li>
		<li><code>disallow</code> - Media Codec(s) to disallow</li>
		<li><code>dtmf_mode</code> - DTMF mode</li>
		<li><code>media_address</code> - IP address used in SDP for media handling</li>
		<li><code>force_rport</code> - Force use of return port</li>
		<li><code>ice_support</code> - Enable the ICE mechanism to help traverse NAT</li>
		<li><code>identify_by</code> - Way(s) for Endpoint to be identified</li>
		<li><code>redirect_method</code> - How redirects received from an endpoint are handled</li>
		<li><code>mailboxes</code> - NOTIFY the endpoint when state changes for any of the specified mailboxes</li>
		<li><code>moh_suggest</code> - Default Music On Hold class</li>
		<li><code>outbound_auth</code> - Authentication object used for outbound requests</li>
		<li><code>outbound_proxy</code> - Proxy through which to send requests, a full SIP URI must be provided</li>
		<li><code>rewrite_contact</code> - Allow Contact header to be rewritten with the source IP address-port</li>
		<li><code>rtp_ipv6</code> - Allow use of IPv6 for RTP traffic</li>
		<li><code>rtp_symmetric</code> - Enforce that RTP must be symmetric</li>
		<li><code>send_diversion</code> - Send the Diversion header, conveying the diversion information to the called user agent</li>
		<li><code>send_pai</code> - Send the P-Asserted-Identity header</li>
		<li><code>send_rpid</code> - Send the Remote-Party-ID header</li>
		<li><code>timers_min_se</code> - Minimum session timers expiration period</li>
		<li><code>timers</code> - Session timers for SIP packets</li>
		<li><code>timers_sess_expires</code> - Maximum session timer expiration period</li>
		<li><code>transport</code> - Desired transport configuration</li>
		<li><code>trust_id_inbound</code> - Accept identification information received from this endpoint</li>
		<li><code>trust_id_outbound</code> - Send private identification details to the endpoint.</li>
		<li><code>type</code> - Must be of type 'endpoint'.</li>
		<li><code>use_ptime</code> - Use Endpoint's requested packetisation interval</li>
		<li><code>use_avpf</code> - Determines whether res_pjsip will use and enforce usage of AVPF for this endpoint.</li>
		<li><code>force_avp</code> - Determines whether res_pjsip will use and enforce usage of AVP, regardless of the RTP profile in use for this endpoint.</li>
		<li><code>media_use_received_transport</code> - Determines whether res_pjsip will use the media transport received in the offer SDP in the corresponding answer SDP.</li>
		<li><code>media_encryption</code> - Determines whether res_pjsip will use and enforce usage of media encryption for this endpoint.</li>
		<li><code>inband_progress</code> - Determines whether chan_pjsip will indicate ringing using inband progress.</li>
		<li><code>call_group</code> - The numeric pickup groups for a channel.</li>
		<li><code>pickup_group</code> - The numeric pickup groups that a channel can pickup.</li>
		<li><code>named_call_group</code> - The named pickup groups for a channel.</li>
		<li><code>named_pickup_group</code> - The named pickup groups that a channel can pickup.</li>
		<li><code>device_state_busy_at</code> - The number of in-use channels which will cause busy to be returned as device state</li>
		<li><code>t38_udptl</code> - Whether T.38 UDPTL support is enabled or not</li>
		<li><code>t38_udptl_ec</code> - T.38 UDPTL error correction method</li>
		<li><code>t38_udptl_maxdatagram</code> - T.38 UDPTL maximum datagram size</li>
		<li><code>fax_detect</code> - Whether CNG tone detection is enabled</li>
		<li><code>t38_udptl_nat</code> - Whether NAT support is enabled on UDPTL sessions</li>
		<li><code>t38_udptl_ipv6</code> - Whether IPv6 is used for UDPTL Sessions</li>
		<li><code>tone_zone</code> - Set which country's indications to use for channels created for this endpoint.</li>
		<li><code>language</code> - Set the default language to use for channels created for this endpoint.</li>
		<li><code>one_touch_recording</code> - Determines whether one-touch recording is allowed for this endpoint.</li>
		<li><code>record_on_feature</code> - The feature to enact when one-touch recording is turned on.</li>
		<li><code>record_off_feature</code> - The feature to enact when one-touch recording is turned off.</li>
		<li><code>rtp_engine</code> - Name of the RTP engine to use for channels created for this endpoint</li>
		<li><code>allow_transfer</code> - Determines whether SIP REFER transfers are allowed for this endpoint</li>
		<li><code>sdp_owner</code> - String placed as the username portion of an SDP origin (o=) line.</li>
		<li><code>sdp_session</code> - String used for the SDP session (s=) line.</li>
		<li><code>tos_audio</code> - DSCP TOS bits for audio streams</li>
		<li><code>tos_video</code> - DSCP TOS bits for video streams</li>
		<li><code>cos_audio</code> - Priority for audio streams</li>
		<li><code>cos_video</code> - Priority for video streams</li>
		<li><code>allow_subscribe</code> - Determines if endpoint is allowed to initiate subscriptions with Asterisk.</li>
		<li><code>sub_min_expiry</code> - The minimum allowed expiry time for subscriptions initiated by the endpoint.</li>
		<li><code>from_user</code> - Username to use in From header for requests to this endpoint.</li>
		<li><code>mwi_from_user</code> - Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</li>
		<li><code>from_domain</code> - Domain to user in From header for requests to this endpoint.</li>
		<li><code>dtls_verify</code> - Verify that the provided peer certificate is valid</li>
		<li><code>dtls_rekey</code> - Interval at which to renegotiate the TLS session and rekey the SRTP session</li>
		<li><code>dtls_cert_file</code> - Path to certificate file to present to peer</li>
		<li><code>dtls_private_key</code> - Path to private key for certificate file</li>
		<li><code>dtls_cipher</code> - Cipher to use for DTLS negotiation</li>
		<li><code>dtls_ca_file</code> - Path to certificate authority certificate</li>
		<li><code>dtls_ca_path</code> - Path to a directory containing certificate authority certificates</li>
		<li><code>dtls_setup</code> - Whether we are willing to accept connections, connect to the other party, or both.</li>
		<li><code>srtp_tag_32</code> - Determines whether 32 byte tags should be used instead of 80 byte tags.</li>
		<li><code>set_var</code> - Variable set on a channel involving the endpoint.</li>
		<li><code>message_context</code> - Context to route incoming MESSAGE requests to.</li>
		<li><code>accountcode</code> - An accountcode to set automatically on any channels created for this endpoint.</li>
	</ul>
	</li>
</ul>


<h3 id="Asterisk13Function_PJSIP_ENDPOINT-SeeAlso">See Also</h3>


<h3 id="Asterisk13Function_PJSIP_ENDPOINT-ImportVersion">Import Version</h3>

<p>This documentation was imported from Asterisk Version SVN-branch-13-r420538</p>
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                    <p>Document generated by Confluence on Aug 11, 2014 13:47</p>
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