jcolp March 15, 2018, 2:52pm #6 IP addresses may have a subnet mask appended. Their traffic will only be coming from 203.0.113.1, Remove all PJSIP modules from the modules directory (often, /usr/lib/asterisk/modules), Remove the configuration file (pjsip.conf). There are several methods to disable or remove modules in Asterisk. Any removed contacts will expire the soonest. This option only applies if media_encryption is set to dtls. This method has some security considerations because an Authentication header is not present on the first message of a dialog when digest authentication is used. Plain text password used for authentication. FreePBX is Asterisk based. Maximum number of threads in the res_pjsip threadpool. Identifier names are usually derived from and can be found in the endpoint identifier module itself (res_pjsip_endpoint_identifier_*). If set to userpass then we'll read from the 'password' option. Including the role of extensions.conf (dialplan) in your overall Asterisk configuration. Usually in Asterisk PJSIP it can happen due to two things. A path to a .crt or .pem file can be provided. direct_media=no. In these cases you will want to consider the below settings for the remote endpoints. At the time of SDP creation, the IP address defined here will be used asthe media address for individual streams in the SDP. See remove_existing and max_contacts for further information about how these 3 settings interact. Determines if endpoint is allowed to initiate subscriptions with Asterisk. Enable/Disable sending unsolicited MWI to all endpoints on startup. The core feature code transfer . At the time of SDP creation, the IP address defined here will be used as the media address for individual streams in the SDP. When the initial unsolicited MWI notification are enabled on startup then the initial notifications get sent at startup. A contact that cannot survive a restart/boot. This option configures the number of seconds without RTP (while on hold) before considering a channel as dead. Some UAs use OPTIONS requests like a 'ping' and the expectation is that they will return a 200 OK. Preferences for selecting codecs for an incoming call. This list will consist of only those codecs found in both lists. This option can be set to send the session to the fax extension when a CNG tone is detected. There is a router interfacing the private and public networks. A way of creating an aliased name to a SIP URI, Authenticates a qualify challenge response if needed, Outbound proxy used when sending OPTIONS request. Number of seconds before an idle thread should be disposed of. For multiple channel variables specify multiple 'set_var'(s). Yeastar S-Series VoIP PBX supports AMI and the default port is 5038 (TCP). If disabled it can improve realtime performance by reducing the number of database requests. direct_media : false. Accept identification information received from this endpoint. This is a string that describes how the codecs specified on an incoming SDP offer (pending) are reconciled with the codecs specified on an endpoint (configured) before being sent to the Asterisk core. If greater than the qualify_frequency for an aor, qualify_frequency will be used instead. Each security mechanism must be in the form defined by RFC 3329 section 2.2. The User-Agent is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually. lordaker March 15, 2018, 2:50pm #5 Ok, make this command so : /etc/init.d/asterisk restart That it ? Best regards, Torbj I'm setup a Asterisk 16.1.1 (endpoints are in realtime), with path support on PJSIP stack. No release has yet been made which contains the linked fix commit. Channel driver technologies such as chan_sip and chan_pjsip have native capability for various transfer types. Evaluate Confluence today. If the contact doesn't respond to the OPTIONS request before the timeout, the contact is marked unavailable. '.' This can be useful for improving compatibility with an ITSP that likes to use user options for whatever reason. app_voicemail mailboxes must be specified as mailbox@context; for example: mailboxes=6001@default. prefer: pending, operation: intersect, keep: all. Options that apply globally to all SIP communications. If this option is set to uri_core the target URI is returned to the dialing application which dials it using the PJSIP channel driver and endpoint originally used. Note that this option is reserved for future functionality. If set to no, res_pjsip will use the AVP or SAVP RTP profile for all media offers on outbound calls and media updates, and will decline media offers not using the AVP or SAVP profile. When a redirect is received from an endpoint there are multiple ways it can be handled. The rest of the options may depend on your particular configuration, phone model, network settings, ITSP, etc. Send RTP back to the same address/port we received it from. Powered by a free Atlassian Confluence Open Source Project License granted to Asterisk Project. If you are seeing messages like: Bridged Calls Direct media is not being used Inbound Registrations Outbound Registrations Inbound Subscriptions If specified, any channel created for this endpoint will automatically have this accountcode set on it. Use the defaults but keep oinly the first codec. The option determines how many seconds into a call before the fax_detect option is disabled for the call. In this post, we'll cover how to use the module, as well as potential avenues for future enhancements to its functionality. Enable sending AMI ContactStatus event when a device refreshes its registration. When Asterisk generates a challenge, the digest realm will be set to this value if there is no better option (such as auth/realm) to be used. MWI taskprocessor high water alert trigger level. Use the same transport for outgoing requests as incoming ones. When your (remote) phone is behind NAT, you may want to check the UDP timeout in your gateway and adjust the "maximum_expiration" time in your phone's AOR settings, like this: If your router/gateway/modem is a Linux device with default settings, the UDP "stream" timeout default is 180, so 160 is a safe setting for your phone to re-register. All versions up to an including 2.11.1 are affected. If this option is set to user the user portion of the redirect target is treated as an extension within the dialplan and dialed using a Local channel. Forwarding this 183 can cause loss of ringback tone. But I can't find options like alwaysauthreject and allowguests in this configuration. When enabled the UDPTL stack will use IPv6. In the above example we assumed the phone was on the same local network as Asterisk. If negotiated this will result in multiple RTP streams being carried over the same underlying transport. Transfer features provided by the Asterisk core are configured in features.conf and accessed with feature codes. Example: If trust_id_inbound is set to yes, the presence of a Privacy: id header in a SIP request or response would indicate the identification provided in the request is private. There are still lots of things to implement and/or test. My config: The key is to make sure you have those three options set appropriately. In various parts of PJSIP, when error/failure occurs, it is found that the function returns without releasing the currently held locks. When a request or response is sent out from Asterisk, if the destination of the message is outside the IP network defined in the option 'local_net', and the media address in the SDP is within the localnet network, then the media address in the SDP will be rewritten to the value defined for 'external_media_address'. If true and a qualify request receives a challenge response then authentication is attempted before declaring the contact available. Condense MWI notifications into a single NOTIFY. This can send a 180 Ringing response before the call has even reached the far end. Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state changes happen for any of the specified mailboxes. The feature designated here can be any built-in or dynamic feature defined in features.conf. If Asterisk is unable to determine which endpoint the SIP request is coming from, then the incoming request will be rejected. You can control how many unmatched requests are received from a single ip address before a security event is generated using the unidentified_request parameters in the "global" configuration object. The remove_existing and remove_unavailable options can help by removing either the soonest to expire or unavailable contact(s) over max_contacts which is likely the old rewrite_contact contact source address being refreshed. Using the same auth section for inbound and outbound authentication is not recommended. If media_address is specified, this option causes the UDPTL instance to be bound to the specified ip address which causes the packets to be sent from that address. Asterisk PJSIP Setting Don't Fragment Bit On UDP; 5s Delays Before Executing The Dialplan; RTP Address Learning And Timing Problem; Asterisk Simply Stops Call Processing; Not Reporting IP Of The Incoming Connection 18.14.0; Github - Mlan; Asterisk Rtp.conf Stunaddr Setting - What Happens If There Is An Outage; Set Codec Based On B Side When configured with chan_sip, peers that are, relative to Asterisk, located behind a NAT are configured using the nat parameter. This option does nothing as we will always complete the challenge response authentication if the qualify request is challenged. Immediately send connected line updates on unanswered incoming calls. When Asterisk sends the INVITE to the SIP trunk, it includes G722 and G729 in the SDP offer (as well as PCMU). Do not perform NAT handling other than RFC 3581. This option determines whether res_pjsip will send private identification information to the endpoint. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. On outbound requests, force the user portion of the Contact header to this value. An Ansible role for installing asterisk. The feature designated here can be any built-in or dynamic feature defined in features.conf. asterisk -- asterisk The multi-part body parser in PJSIP, as used in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1, Certified Asterisk 13.13 before 13.13-cert4, and other products, allows remote attackers to cause a denial of service (out-of-bounds read and application crash) via a crafted packet. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. In the pjsip channel driver (res_pjsip) in Asterisk 13.x before 13.17.1 and 14.x before 14.6.1, a carefully crafted tel URI in a From, To, or Contact . This examples shows the configuration required for: This shows configuration for a SIP trunk as would typically be provided by an ITSP. The string actually specifies 4 name:value pair parameters separated by commas. This is much like the external_media_address setting, but for SIP signaling instead of RTP media. Set which country's indications to use for channels created for this endpoint. (typically /etc/asterisk/). The rewrite_contact option registers the source address as the contact address to help with NAT and reusing connection oriented transports such as TCP and TLS. More information about these options can be found on the . celsoannes August 21, 2019, 5:28pm #12 Thanks for the clarification. This is the IP network that we want to consider our local network. Determines whether new contacts replace existing ones. See link for more: http://www.openssl.org/docs/apps/ciphers.html#CIPHER\_SUITE\_NAMES. Whether we are willing to accept connections, connect to the other party, or both. Use a separate "contact=" entry for each contact required. Codec negotiation prefs for incoming offers. Type of hash to use for the DTLS fingerprint in the SDP. When a request or response is sent out, if the destination of the message is outside the IP network defined in the option localnet, and the media address in the SDP is within the localnet network, then the media address in the SDP will be rewritten to the value defined for external_media_address. Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK" after REFER has been accepted. If no message_context is specified, then the context setting is used. When Asterisk generates an outgoing SIP request, the From header username will be set to this value if there is no better option (such as CallerID) to be used. Always check your logs for warnings or errors if you suspect something is wrong. However, to allow anonymous calls you need to create an endpoint named "anonymous" (or any of the variants listed below if the disable_multi_domain option is 'no') and load res_pjsip_endpoint_identifier_anonymous.so. You understand basic Asterisk concepts. make[3]: Entering directory '/build/lede-17.01-phase2/mips64el_mips64/build/sdk/feeds/telephony/net/asterisk-13.x' rm -f /build/lede-17.01-phase2/mips64el_mips64 . This is important, because our Asterisk system has a private IP address that the ITSP cannot route to. This effectively makes the semicolon a non-usable character for PJSIP endpoint names, extensions, and AORs. NOTE: Be aware that the 'external_media_address' option, set in Transportconfiguration, can also affect the final media address used in the SDP. And if not, why was this left out? This method of identification has some security considerations because an Authentication header is not present on the first message of a dialog when digest authentication is used. @jcolp I install it by following the process in the wiki Asterisk and its work Thanks, Powered by Discourse, best viewed with JavaScript enabled, https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip. Context to route incoming MESSAGE requests to. To insure that the script can read any #include'd files, run it from the /etc/asterisk directory or in another location with a copy of the sip.conf and any included files. PJSIP is the new channel library for Asterisk, replacing the older DAHDI and LIBPRI drivers. direct_media_method : invite. Time in seconds. In versions 1.8 and greater of Asterisk, the following nat parameter options are available: Versions of Asterisk prior to 1.8 had less granularity for the nat parameter: In chan_pjsip, theendpoint options that control NAT behavior are: In the pjsip trunk configuration shouldn't the server_uri be the provider's IP and the client_uri my IP?