two SIP phones need to make calls to or through Asterisk, we also want to be able to call them from Asterisk, for them to be identified as users (in the old chan_sip) or endpoints (in the new res_sip/chan_pjsip), both devices need to use username and password authentication, 6001 is setup to allow registration to Asterisk, and 6002 is setup with a static host/contact, SIP provider requires registration to their server with a username of "myaccountname" and a password of "1234567890", SIP provider requires registration to their server at the address of 203.0.113.1:5060. 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. This option only applies if media_encryption is set to dtls. prefer: pending, operation: intersect, keep: all. This should be set to yes and max_contacts set to 1 if you wish to stick with the older chan_sip behaviour. Coming in Asterisk 13.8.0, a new module - res_pjsip_history - has been added that provides capturing, filtering, and display of SIP messages. Use Endpoint's requested packetization interval. If 0 no timeout. No transcoding allowed. You understand basic Asterisk concepts. When enabled the UDPTL stack will use IPv6. Are you telling me that I am sending to the provider my IP so he can route the calls where I ask?I am still confused about the difference between the server_uri and client_uri A SIP REGISTER is for telling a remote server where you can be reached. it is adding the following lines: 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. Now the packet capture shows how the media goes through the asterisk interface. When enabled, aggregate_mwi condenses message waiting notifications from multiple mailboxes into a single NOTIFY. Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. If disabled it can improve realtime performance by reducing the number of database requests. With this option enabled, Asterisk will attempt to negotiate the use of bundle. Under certain conditions they could make things worse. This is where you'll be configuring everything related to your inbound or outbound SIP accounts and endpoints. @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. 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). This is a comma-delimited list of security mechanisms to use. On incoming INVITEs, the Identity header will be checked for validity. If set to yes, res_pjsip will use the received media transport. The numeric pickup groups that a channel can pickup. We are assuming you have already read the Configuring res_pjsip page and have a basic understanding of Asterisk. This option only applies if media_encryption is set to dtls. Asterisk and the phones are on a private network. Options that apply globally to all SIP communications. Use only the ones that are common. Enable STIR/SHAKEN support on this endpoint. As an alternative to specifying a plain text password, you can hash the username, realm and password together one time and place the hash value here. That is registration to a remote server, authentication to it and a peer/endpoint setup to allow inbound calls from the provider. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. IP-address of the last Via header from registration. On outgoing calls, if the UAS responds with different SDP attributes on subsequent 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is different than that on the previous one, follow it. Always check your logs for warnings or errors if you suspect something is wrong. If not specified, the context configured for the endpoint will be used. 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. direct_media_glare_mitigation : none. The functionality was written to be familiar to users of chan_sip by allowing it to be . On outgoing calls, if the UAS responds with different SDP attributes on non-100rel 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is the same as that on the previous one, process the updated SDP. The last Via header should contain the address of UA which sent the request. If any taskprocessor queue size reaches its high water level then pjsip will stop processing new requests until the alert is cleared. set in pjsip.endpoint.conf. The two external* options mentioned here should be set to the same address unless you separate your signaling and media to different addresses or servers. 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. At the time of SDP creation, the IP address defined here will be used asthe media address for individual streams in the SDP. Set to -1 for the low water level to be 90% of the high water level. These option is for chan_sip not needed on pjsip, also you dont need an aor section for anoymous calls. For endpoints that cannot SUBSCRIBE for MWI, you can set the mailboxes option in your endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint. If greater than the qualify_frequency for an aor, qualify_frequency will be used instead. There are several methods to disable or remove modules in Asterisk. Evaluate Confluence today. As well, names only match against a single level meaning '.example.com' matches 'foo.example.com', but not 'foo.bar.example.com'. The uri_pjsip option has the benefit of being more efficient and also supporting multiple potential redirect targets. Evaluate Confluence today. The trunk seems to always negotiate to G729, so Asterisk ends up transcoding the ulaw to G729 between the two, and faxes have lots of issues. In that case, it is best to disable res_pjsip unless you understand how to configure them both together. PJSIP will not automatically switch the sending one to the receiving one. [CDATA[*/ I reload the module in the Asterisk CLI too by this command : Noload only tells Asterisk at load time not to load chan_sip. This option must also be enabled on endpoints that require this functionality. Name of the RTP engine to use for channels created for this endpoint, Determines whether SIP REFER transfers are allowed for this endpoint, Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number, Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side. How can I configure static IP for chan_pjsip extensions? By default this option is set to 0, which means do not check. If this is not set or the value provided is 0 rekeying will be disabled. String style specification. Default. Username to use in From header for unsolicited MWI NOTIFYs to this endpoint. See the auth realm description for details. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This will force the endpoint to use the specified transport configuration to send SIP messages. This option allows the 'Q.850' Reason header to be suppressed. This is a comma-delimited list of auth sections defined in pjsip.conf to be used to verify inbound connection attempts. If your Asterisk PBX is behind a NAT firewall, i.e. You can use the CLI command "pjsip show identifiers" to see the identifiers currently available. SIP-. On outgoing calls, if the UAS responds with different SDP attributes on subsequent 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is different than that on the previous one, follow it. 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 can't use pre-hashed passwords with a wildcard auth object. Whether we are willing to accept connections, connect to the other party, or both. This should be set to 1 and remove_existing set to yes if you wish to stick with the older chan_sip behaviour. For more information on this timer, see RFC 3261, Section 17.1.1.1. This is the IP network that we want to consider our local network. A variety of reference content is provided in the following sub-pages. Enable/Disable sending unsolicited MWI to all endpoints on startup. For the sake of a complete example and clarity, in this example we use the following fake details: DID number provided by ITSP: 19998887777. 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 . The migration script is just that, a handy script to migrate if you have an existing sip.conf and dont want to start from scratch. An accountcode to set automatically on any channels created for this endpoint. Including the role of extensions.conf (dialplan) in your overall Asterisk configuration. Forwarding this 183 can cause loss of ringback tone. This option only applies if media_encryption is set to sdes or dtls. 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. This geolocation profile will be applied to all calls received by the channel driver from the dialplan before they're forwarded the remote endpoint. If specified, incoming MESSAGE requests will be routed to the indicated dialplan context. Disabling res_pjsip and chan_pjsip You may want to keep using chan_sip for a short time in Asterisk 12+ while you migrate to res_pjsip. IP address used in SDP for media handling. Set transaction timer T1 value (milliseconds). 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 The feature designated here can be any built-in or dynamic feature defined in features.conf. On outgoing INVITEs, an Identity header will be added. Note that this option is reserved for future functionality. Control whether dialog-info subscriptions get 'early' state on Ringing when already INUSE. If I set inband_progress = no in pjsip.conf, Asterisk will still send a Session Progress to the caller, which if I remember correctly corresponds to setting progressinband=no i sip.conf. Value used in Max-Forwards header for SIP requests. Thanks for . If Asterisk is already running you can unload chan_sip using module unload chan_sip.so from the console, but if it started before PJSIP then it would cause problems. If your UDP stream timeout is larger (/proc/sys/net/netfilter/nf_conntrack_udp_timeout_stream), you may adjust maximum_expiration accordingly. , . The order by which endpoint identifiers are processed and checked. prefer: pending, operation: intersect, keep: all, transcode: allow. When an INFO request for one-touch recording arrives with a Record header set to "on", this feature will be enabled for the channel. If Asterisk is already running you can unload chan_sip using "module unload chan_sip.so" from the console, but if it started before PJSIP then it would cause problems. The following values are valid: This setting only describes whether the password is in plain text or has been pre-hashed with MD5. At this time, the only part of Asterisk that uses sorcery for configuration is PJSIP. If set to yes, chan_pjsip will send a 183 Session Progress when told to indicate ringing and will immediately start sending ringing as audio. Must be of type 'system' UNLESS the object name is 'system'. Follow SDP forked media when To tag is the same. Setting the value to zero disables the timeout. If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted and relayed. There is a router interfacing the private and public networks. Settings > Asterisk Settings . If remove_existing is set to yes, setting remove_unavailable to yes will prioritize unavailable contacts for removal instead of just removing the contact that expires the soonest. Just remove the --libdir=/usr/lib64 option from the command. But sometimes FreePBX is disabling my pjsip modules at startup by modifying the modules.conf. Together these options make sure the far end knows where to send back SIP and RTP packets, and direct_media ensures Asterisk stays in the media path. This option applies both to calls originating from the endpoint and calls originating from Asterisk. This setting attempts to avoid creating INVITE glare scenarios by disabling direct media reINVITEs in one direction thereby allowing designated servers (according to this option) to initiate direct media reINVITEs without contention and significantly reducing call setup time. Allow transcoding. When the initial unsolicited MWI notification are enabled on startup then the initial notifications get sent at startup. In that case, it is best to disable res_pjsip unless you understand how to configure them both together. Use the same transport for outgoing requests as incoming ones. Disable automatic switching from UDP to TCP transports if outgoing request is too large. The option determines how many seconds into a call before the fax_detect option is disabled for the call. The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor, Enable/Disable SIP debug logging. If no, private Caller-ID information will not be forwarded to the endpoint. This examples shows the configuration required for: This shows configuration for a SIP trunk as would typically be provided by an ITSP. Remove "rport" parameter from the outgoing requests. 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. 2017-06-02: not yet calculated IBM X-Force ID: 126873. This option enforces a limit on the maximum simultaneous negotiated video streams allowed for the endpoint. SIP provider requires outbound calls to their server at the same address of registration, plus using same authentication details. Minimum session timer expiration period. Whitespace is ignored and they may be specified in any order. 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 . UDP). 'f.example.com' and 'foo..com' are not allowed. The default input file is sip.conf, and the default output file is pjsip.conf. When the initial unsolicited MWI notifications are disabled on startup then the notifications will start on the endpoint's next contact update. You have installed pjproject, a dependency for res_pjsip. Powered by a free Atlassian Confluence Open Source Project License granted to Asterisk Project. There are security implications to enabling this setting as it can allow information disclosure to occur - specifically, if enabled, an external party could enumerate and find the endpoint name by sending OPTIONS requests and examining the responses. When configured with chan_sip, peers that are, relative to Asterisk, located behind a NAT are configured using the nat parameter. Allow the sending and receiving RTP codec to differ, Enable RFC 5761 RTCP multiplexing on the RTP port, Whether to notifies all the progress details on blind transfer, Whether to notifies dialog-info 'early' on InUse&Ringing state, The maximum number of allowed audio streams for the endpoint, The maximum number of allowed video streams for the endpoint, Defaults and enables some options that are relevant to WebRTC, Mailbox name to use when incoming MWI NOTIFYs are received, Follow SDP forked media when To tag is different, Accept multiple SDP answers on non-100rel responses, Suppress Q.850 Reason headers for this endpoint, Do not forward 183 when it doesn't contain SDP, Enable STIR/SHAKEN support on this endpoint, STIR/SHAKEN profile containing additional configuration options, Skip authentication when receiving OPTIONS requests. Authentication Object(s) associated with the endpoint, Mitigation of direct media (re)INVITE glare, Accept Connected Line updates from this endpoint, Send Connected Line updates to this endpoint. When an INFO request for one-touch recording arrives with a Record header set to "off", this feature will be enabled for the channel. This option determines whether res_pjsip will send private identification information to the endpoint. This value does not affect the number of contacts that can be added with the "contact" option. Allow use of wildcards in certificates (TLS ONLY). 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. Including the role of extensions.conf (dialplan) in your overall Asterisk configuration. Number of seconds before an idle thread should be disposed of. For now, understand that it is a CRUD (create, read, update, delete) API in Asterisk that can read and write to different backends. The configuration for a location of an endpoint. Evaluate Confluence today. And if not, why was this left out? This will result in RTP and RTCP being sent and received on the same port. Determines whether encryption should be used if possible but does not terminate the session if not achieved. 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. RFC 3261 specifies this as a SHOULD requirement. When Asterisk sends the INVITE to the SIP trunk, it includes G722 and G729 in the SDP offer (as well as PCMU). See link for more: http://www.openssl.org/docs/apps/ciphers.html#CIPHER\_STRINGS. Determines whether media may flow directly between endpoints. This is the external IP address to use in RTP handling. A path to a key file can be provided. Set transaction timer B value (milliseconds). Preferences for selecting codecs for an incoming call. Default expiration time in seconds for contacts that are dynamically bound to an AoR. Allow support for RFC3262 provisional ACK tags. The router is performing Network Address Translation and Firewall functions. PJSIP is the new channel library for Asterisk, replacing the older DAHDI and LIBPRI drivers. More than one mailbox can be specified with a comma-delimited string. No. Time in seconds. It depends on how the remote side is set up. I see both "type=" and "type = " (so with and without a space around the equal signs). In these cases you will want to consider the below settings for the remote endpoints. Can be set to a comma separated list of case sensitive strings limited by supported line length. Network to consider local (used for NAT purposes). To configure Asterisk's PJSIP-based SIP channel driver, included with Asterisk versions 12, 13 and newer, to work with Digium's SIP Trunking service, you should configure 6 objects: transport auth aor endpoint registration identify But I can't find options like alwaysauthreject and allowguests in this configuration. It is recommended that this be set to 64 * Timer T1, but it may be set higher if desired. This usually happens when the INVITE is forked to multiple UASs and more than one sends an SDP answer. If a websocket connection accepts input slowly, the timeout for writes to it can be increased to keep it from being disconnected. In old sip server, we were using the following command in AGI. IP addresses may have a subnet mask appended. This is a string that describes how the codecs that come from the core (pending) are reconciled with the codecs specified on an endpoint (configured) when sending an SDP answer. jcolp March 15, 2018, 2:52pm #6 This option can be set to override the maximum datagram of a remote endpoint for broken endpoints. Protocol Behavior 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. a migration by using the script in source folder sip_to_pjsip.py Force g.726 to use AAL2 packing order when negotiating g.726 audio. cc. Merge them with the codecs from the core keeping the order of the preferred list. Resolve the server_uri to an IP address and port, Send a REGISTER request to the IP address and port. Disable automatic switching from UDP to TCP transports. MWI taskprocessor low water clear alert level. Determines if endpoint is allowed to initiate subscriptions with Asterisk. It should be noted that external_media_address and external_signaling_address currently do only allow for IPs as parameter until Asterisk 14.6 and 13.17.Once Asterisk 14.7 and 13.8 are released, this patch herehttps://gerrit.asterisk.org/#/c/6070/should allow for dynamic hosts as parameter. Maximum time to keep a peer with explicit expiration. Un-install and re-install Asterisk with no PJSIP related modules. Codec negotiation prefs for outgoing answers. pkirkham January 29, 2019, 2:36pm 15 It allows live monitoring of events that occur in the system, as well enabling you to request that Asterisk performs some action. I'm using res_pjsip, the configuration is stored in pjsip.conf.

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