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. You may want to keep using chan_sip for a short time in Asterisk 12+ while you migrate to res_pjsip. If set to google_oauth then we'll read from the refresh_token/oauth_clientid/oauth_secret fields. Determines whether media may flow directly between endpoints. As well youll want to ensure that chan_sip.so isnt loaded by adding a noload => chan_sip.so line to modules.conf, [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip, So when I add this line in the modules.conf. The string actually specifies 4 name:value pair parameters separated by commas. As shown in picture, changing NAT = yes and IP Configuration to static in Settings > SIP Settings > Chan SIP Settings solved the issue for chain_sip extensions. The value is a comma-delimited list of IP addresses. Number of seconds between RTP comfort noise keepalive packets. If you are seeing messages like: Bridged Calls Direct media is not being used Inbound Registrations Outbound Registrations Inbound Subscriptions You have Installed Asterisk including the res_pjsip and chan_pjsip modules and their dependencies. IP-port of the last Via header from registration. Currently, only mediasec is supported. And I make These examples contain only the configuration required for sip.conf/pjsip.conf as the configuration for other files should be the same, excepting the Dial statements in your extensions.conf. Thanks for . Direct Media 100rel/early media Re-invites Fax Multi-stream This option is a comma separated list of methods the endpoint can be identified. Do not perform NAT handling other than RFC 3581. since I'm not able to organically reproduce the bug, to test it you can disable pjsip by hand: From FreePBX interface, open "Settings" > "Advanced Settings" find "SIP Channel Driver" variable and set it to "chan_sip" Submit and apply changes Now you should be able to verify the bug condition with grep pjsip /etc/asterisk/modules.conf prefer: pending, operation: intersect, keep: all. , . I'm not sure I got that right. On receiving a new registration to the AoR should it remove enough existing contacts not added or updated by the registration to satisfy max_contacts? I dont know how you have installed Asterisk, so I cant say for certain but that may work. Dialplan context to use for overlap dialing extension matching. This option controls both how an endpoint is matched for incoming traffic and also how an AOR is determined if a registration occurs. All inbound SIP traffic to Asterisk must be matched to a configured endpoint. If media_address is specified, this option causes the RTP instance to be bound to the specified ip address which causes the packets to be sent from that address. SIP provider will call your server with a user name of "mytrunk". 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. The configuration for a location of an endpoint. Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state changes happen for any of the specified mailboxes. The feature to enact when one-touch recording is turned on. Use only the ones that are common. This is important, because our Asterisk system has a private IP address that the ITSP cannot route to. Force g.726 to use AAL2 packing order when negotiating g.726 audio. This option can be set to override the maximum datagram of a remote endpoint for broken endpoints. The maximum amount of time from startup that qualifies should be attempted on all contacts. A contact that cannot survive a restart/boot. Prefer the codecs coming from the caller. Where the public network is the Internet. This should work ;;anoymous calls ;;anonymous [transport-udp-anonymous] type=transport protocol=udp bind=0.0.0.0:5067 [anonymous] type=endpoint context=from-anonymous disallow=all allow=ulaw transport=transport-udp-anonymous Contained within a download of Asterisk, there is a Python script, sip_to_pjsip.py, found within the contrib/scripts/sip_to_pjsip subdirectory, that provides a basic conversion of a sip.conf config to a pjsip.conf config. 09:53:56 AM [Edward] Alternatively you can disable the session timer 09:54:19 AM [Stewart] So the problem is a configuration issue with . 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. Maximum number of seconds without receiving RTP (while on hold) before terminating call. Note the '-n'. Our customer can set up calls to either PSTN or Sip endpoints. I'm using chan_pjsip trunks so I'll try to find where to add the "session-timers=refuse" in the trunk configuration, or I'll change the trunk to chan_sip. The "none" and "pjsip_only" options should be used with extreme caution and only to mitigate specific issues. The other options may be different depending on how you want to use Asterisk. This must be in CIDR or dotted decimal format with the IP and mask separated with a slash ('/'). No release has yet been made which contains the linked fix commit. This value does not affect the number of contacts that can be added with the "contact" option. On outgoing INVITEs, an Identity header will be added. 2017-08-28: not yet calculated: CVE-2017-1376 . 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. 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. The certificate file can be reloaded if the filename in configuration remains unchanged. Evaluate Confluence today. If set the provided URI will be used as the outbound proxy when an OPTIONS request is sent to a contact for qualify purposes. Disable automatic switching from UDP to TCP transports if outgoing request is too large. This option has been deprecated in favor of incoming_call_offer_pref. If no, the configured Caller-ID from pjsip.conf will always be used as the identity for the endpoint. Powered by a free Atlassian Confluence Open Source Project License granted to Asterisk Project. Configuring res_pjsip to work through NAT. 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. Based on this setting, a joint list of preferred codecs between those received in an incoming SDP offer (remote), and those specified in the endpoint's "allow" parameter (local) es created and is passed to the Asterisk core. Issue to setup a HT813 ATA in a pstn line and an Asterisk PBX 13 with PJSIP and Realtime behind NAT, when I call to pstn lines the call is not forwarded to the extension that should Invites arriving in Asterisk CLI console: [Jan 16 12:05:53] NOTICE[32270]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '<sip:019976401569@54.236.1.32>' failed for '201.75.25.1:28140 . app_voicemail mailboxes must be specified as [emailprotected]; for example: [emailprotected] For mailboxes provided by external sources, such as through the res_mwi_external module, you must specify strings supported by the external system. Since Asterisk normally sends a security event when an incoming request can't be matched to an endpoint, using this method requires that the security event be deferred until a request is received with the Authentication header and only generated if the username doesn't result in a match. On incoming INVITEs, the Identity header will be checked for validity. This page and its sub-pages are intended to help an administrator configure the new SIP resources and channel driver included with Asterisk 12. At the specified interval, Asterisk will send an RTP comfort noise frame. Geolocation profile to apply to incoming calls, Geolocation profile to apply to outgoing calls. Type of hash to use for the DTLS fingerprint in the SDP. Use a separate "contact=" entry for each contact required. The private key file can be reloaded if the filename in configuration remains unchanged. Determines whether chan_pjsip will indicate ringing using inband progress. The core feature code transfer . The interval at which unidentified requests are older than twice the unidentified_request_period are pruned. Using the same auth section for inbound and outbound authentication is not recommended. 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. If negotiated this will result in multiple RTP streams being carried over the same underlying transport. Use the CLI command pjsip list ciphers to see a list of cipher names available for your installation. See link for more: http://www.openssl.org/docs/apps/ciphers.html#CIPHER\_SUITE\_NAMES. Timer T1 is the base for determining how long to wait before retransmitting requests that receive no response when using an unreliable transport (e.g. PJSIP is the new channel library for Asterisk, replacing the older DAHDI and LIBPRI drivers. This setting allows to choose the DTMF mode for endpoint communication. in certs for common,and subject alt names of type DNS for TLS transport types. You have Installed Asterisk including the res_pjsip and chan_pjsip modules (implying you installed their dependencies as well) You understand basic Asterisk concepts. This flag emulates the behavior of chan_sip and prevents these 183 responses from being forwarded. Determines whether media may flow directly between endpoints. The effect of this setting depends on the setting of remove_existing. This configuration documentation is for functionality provided by res_pjsip. The client_uri is the URI that tells the server what we want to register to. See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings.