1 ; PJSIP Configuration Samples and Quick Reference
3 ; This file has several very basic configuration examples, to serve as a quick
4 ; reference to jog your memory when you need to write up a new configuration.
5 ; It is not intended to teach PJSIP configuration or serve as an exhaustive
6 ; reference of options and potential scenarios.
8 ; This file has two main sections.
9 ; First, manually written examples to serve as a handy reference.
10 ; Second, a list of all possible PJSIP config options by section. This is
11 ; pulled from the XML config help. It only shows the synopsis for every item.
12 ; If you want to see more detail please check the documentation sources
13 ; mentioned at the top of this file.
17 ; The official documentation is at http://wiki.asterisk.org
18 ; You can read the XML configuration help via Asterisk command line with
19 ; "config show help res_pjsip", then you can drill down through the various
20 ; sections and their options.
23 ;========!!!!!!!!!!!!!!!!!!! SECURITY NOTICE !!!!!!!!!!!!!!!!!!!!===========
25 ; At a minimum please read the file "README-SERIOUSLY.bestpractices.txt",
26 ; located in the Asterisk source directory before starting Asterisk.
27 ; Otherwise you risk allowing the security of the Asterisk system to be
28 ; compromised. Beyond that please visit and read the security information on
29 ; the wiki at: https://wiki.asterisk.org/wiki/x/EwFB
31 ; A few basics to pay attention to:
35 ; By default anonymous inbound calls via PJSIP are not allowed. If you want to
36 ; route anonymous calls you'll need to define an endpoint named "anonymous".
37 ; res_pjsip_endpoint_identifier_anonymous.so handles that functionality so it
38 ; must be loaded. It is not recommended to accept anonymous calls.
40 ; Access Control Lists
42 ; See the example ACL configuration in this file. Read the configuration help
43 ; for the section and all of its options. Look over the samples in acl.conf
44 ; and documentation at https://wiki.asterisk.org/wiki/x/uA80AQ
45 ; If possible, restrict access to only networks and addresses you trust.
49 ; When defining configuration (such as an endpoint) that links into
50 ; dialplan configuration, be aware of what that dialplan does. It's easy to
51 ; accidentally provide access to internal or outbound dialing extensions which
52 ; could cost you severely. The "context=" line in endpoint configuration
53 ; determines which dialplan context inbound calls will enter into.
55 ;=============================================================================
57 ; Overview of Configuration Section Types Used in the Examples
59 ; * Transport "transport"
60 ; * Configures res_pjsip transport layer interaction.
61 ; * Endpoint "endpoint"
62 ; * Configures core SIP functionality related to SIP endpoints.
63 ; * Authentication "auth"
64 ; * Stores inbound or outbound authentication credentials for use by trunks,
65 ; endpoints, registrations.
66 ; * Address of Record "aor"
67 ; * Stores contact information for use by endpoints.
68 ; * Endpoint Identification "identify"
69 ; * Maps a host directly to an endpoint
70 ; * Access Control List "acl"
71 ; * Defines a permission list or references one stored in acl.conf
72 ; * Registration "registration"
73 ; * Contains information about an outbound SIP registration
75 ; The following sections show example configurations for various scenarios.
76 ; Most require a couple or more configuration types configured in concert.
78 ;=============================================================================
80 ; Naming of Configuration Sections
82 ; Configuration section names are denoted with enclosing brackets,
84 ; In most cases, you can name a section whatever makes sense to you. For example
85 ; you might name a transport [transport-udp-nat] to help you remember how that
86 ; section is being used. However, in some cases, ("endpoint" and "aor" types)
87 ; the section name has a relationship to its function.
89 ; Depending on the modules loaded, Asterisk can match SIP requests to an
90 ; endpoint or aor in a few ways:
92 ; 1) Match a section name for endpoint type sections to the username in the
93 ; "From" header of inbound SIP requests.
94 ; 2) Match a section name for aor type sections to the username in the "To"
95 ; header of inbound SIP REGISTER requests.
96 ; 3) With an identify type section configured, match an inbound SIP request of
97 ; any type to an endpoint or aor based on the IP source address of the
100 ; Note that sections can have the same name as long as their "type" options are
101 ; set to different values. In most cases it makes sense to have associated
102 ; configuration sections use the same name, as you'll see in the examples within
105 ;===============EXAMPLE TRANSPORTS============================================
107 ; A few examples for potential transport options.
109 ; For the NAT transport example, be aware that the options starting with
110 ; the prefix "external_" will only apply to communication with addresses
111 ; outside the range set with "local_net=".
113 ; IPv6: For endpoints using IPv6, remember to set "rtp_ipv6=yes" so that the RTP
114 ; engine will also be able to bind to an IPv6 address.
116 ; You can have more than one of any type of transport, as long as it doesn't
117 ; use the same resources (bind address, port, etc) as the others.
119 ; Basic UDP transport
123 ;protocol=udp ;udp,tcp,tls,ws,wss
126 ; UDP transport behind NAT
132 ;local_net=192.0.2.0/24
133 ;external_media_address=203.0.113.1
134 ;external_signaling_address=203.0.113.1
136 ; Basic IPv6 UDP transport
138 ;[transport-udp-ipv6]
143 ; Example IPv4 TLS transport
149 ;cert_file=/path/mycert.crt
150 ;privkey_file=/path/mykey.key
155 ;===============OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION============
157 ; This is a simple registration that works with some SIP trunking providers.
158 ; You'll need to set up the auth example "mytrunk_auth" below to enable outbound
159 ; authentication. Note that we "outbound_auth=" use for outbound authentication
160 ; instead of "auth=", which is for inbound authentication.
162 ; If you are registering to a server from behind NAT, be sure you assign a transport
163 ; that is appropriately configured with NAT related settings. See the NAT transport example.
165 ; "contact_user=" sets the SIP contact header's user portion of the SIP URI
166 ; this will affect the extension reached in dialplan when the far end calls you at this
167 ; registration. The default is 's'.
171 ;transport=transport-udp
172 ;outbound_auth=mytrunk_auth
173 ;server_uri=sip:sip.example.com
174 ;client_uri=sip:1234567890@sip.example.com
175 ;contact_user=1234567890
177 ;forbidden_retry_interval=600
185 ;realm=sip.example.com
187 ;===============ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION=======
189 ; This is one way to configure an endpoint as a trunk. It is set up with
190 ; "outbound_auth=" to enable authentication when dialing out through this
191 ; endpoint. There is no inbound authentication set up since a provider will
192 ; not normally authenticate when calling you.
194 ; The identify configuration enables IP address matching against this endpoint.
195 ; For calls from a trunking provider, the From user may be different every time,
196 ; so we want to match against IP address instead of From user.
198 ; If you want the provider of your trunk to know where to send your calls
199 ; you'll need to use an outbound registration as in the example above this
204 ; At a basic level configure the endpoint with a transport that is set up
205 ; with the appropriate NAT settings. There may be some additional settings you
206 ; need here based on your NAT/Firewall scenario. Look to the CLI config help
207 ; "config show help res_pjsip endpoint" or on the wiki for other NAT related
208 ; options and configuration. We've included a few below.
212 ; Endpoints use one or more AOR sections to store their contact details.
213 ; You can define multiple contact addresses in SIP URI format in multiple
214 ; "contact=" entries.
219 ;transport=transport-udp
220 ;context=from-external
223 ;outbound_auth=mytrunk_auth
225 ; ;A few NAT relevant options that may come in handy.
226 ;force_rport=yes ;It's a good idea to read the configuration help for each
227 ;direct_media=no ;of these options.
232 ;contact=sip:198.51.100.1:5060
233 ;contact=sip:198.51.100.2:5060
242 ;=============ENDPOINT CONFIGURED AS A TRUNK, INBOUND AUTH AND REGISTRATION===
244 ; Here we are allowing a remote device to register to Asterisk and requiring
245 ; that they authenticate for registration and calls.
246 ; You'll note that this configuration is essentially the same as configuring
247 ; an endpoint for use with a SIP phone.
252 ;context=from-external
255 ;transport=transport-udp
270 ;===============ENDPOINT CONFIGURED FOR USE WITH A SIP PHONE==================
272 ; This example includes the endpoint, auth and aor configurations. It
273 ; requires inbound authentication and allows registration, as well as references
274 ; a transport that you'll need to uncomment from the previous examples.
276 ; Uncomment one of the transport lines to choose which transport you want. If
277 ; not specified then the default transport chosen is the first defined transport
278 ; in the configuration file.
280 ; Modify the "max_contacts=" line to change how many unique registrations to allow.
282 ; Use the "contact=" line instead of max_contacts= if you want to statically
283 ; define the location of the device.
285 ; If using the TLS enabled transport, you may want the "media_encryption=yes"
286 ; option to additionally enable SRTP, though they are not mutually inclusive.
288 ; Use the "rtp_ipv6=yes" option if you want to utilize RTP over an ipv6 transport.
290 ; If this endpoint were remote, and it was using a transport configured for NAT
291 ; then you likely want to use "direct_media=no" to prevent audio issues.
296 ;transport=transport-udp
297 ;context=from-internal
304 ; A few more transports to pick from, and some related options below them.
306 ;transport=transport-tls
307 ;media_encryption=yes
308 ;transport=transport-udp-ipv6
310 ;transport=transport-udp-nat
313 ; MWI related options
316 ;mailboxes=6001@default,7001@default
319 ; Extension and Device state options
321 ;device_state_busy_at=1
334 ;contact=sip:6001@192.0.2.1:5060
336 ;===============ENDPOINT BEHIND NAT OR FIREWALL===============================
338 ; This example assumes your transport is configured with a public IP and the
339 ; endpoint itself is behind NAT and maybe a firewall, rather than having
340 ; Asterisk behind NAT. For the sake of simplicity, we'll assume a typical
341 ; VOIP phone. The most important settings to configure are:
343 ; * direct_media, to ensure Asterisk stays in the media path
344 ; * rtp_symmetric and force_rport options to help the far-end NAT/firewall
346 ; Depending on the settings of your remote SIP device or NAT/firewall device
347 ; you may have to experiment with a combination of these settings.
349 ; If both Asterisk and the remote phones are a behind NAT/firewall then you'll
350 ; have to make sure to use a transport with appropriate settings (as in the
351 ; transport-udp-nat example).
355 ;transport=transport-udp
356 ;context=from-internal
364 ;ice_support=yes ;This is specific to clients that support NAT traversal
365 ;for media via ICE,STUN,TURN. See the wiki at:
366 ;https://wiki.asterisk.org/wiki/x/D4FHAQ
367 ;for a deeper explanation of this topic.
380 ;============EXAMPLE ACL CONFIGURATION==========================================
382 ; The ACL or Access Control List section defines a set of permissions to permit
383 ; or deny access to various address or addresses. Alternatively it references an
384 ; ACL configuration already set in acl.conf.
386 ; The ACL configuration is independent of individual endpoint configuration and
387 ; operates on all inbound SIP communication using res_pjsip.
389 ; Reference an ACL defined in acl.conf.
393 ;acl=example_named_acl1
395 ; Reference a contactacl specifically.
399 ;contact_acl=example_contact_acl1
401 ; Define your own ACL here in pjsip.conf and
402 ; permit or deny by IP address or range.
406 ;deny=0.0.0.0/0.0.0.0
407 ;permit=209.16.236.0/24
410 ; Restrict based on Contact Headers rather than IP.
411 ; Define options multiple times for various addresses or use a comma-delimited string.
415 ;contact_deny=0.0.0.0/0.0.0.0
416 ;contact_permit=209.16.236.0/24
417 ;contact_permit=209.16.236.1
418 ;contact_permit=209.16.236.2,209.16.236.3
420 ; Restrict based on Contact Headers rather than IP and use
421 ; advanced syntax. Note the bang symbol used for "NOT", so we can deny
422 ; 209.16.236.12/32 within the permit= statement.
426 ;contact_deny=0.0.0.0/0.0.0.0
427 ;contact_permit=209.16.236.0
428 ;permit=209.16.236.0/24, !209.16.236.12/32
432 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip
433 ;==========================ENDPOINT SECTION OPTIONS=========================
436 ;100rel=yes ; Allow support for RFC3262 provisional ACK tags (default:
438 ;aggregate_mwi=yes ; (default: "yes")
439 ;allow= ; Media Codec s to allow (default: "")
440 ;aors= ; AoR s to be used with the endpoint (default: "")
441 ;auth= ; Authentication Object s associated with the endpoint (default: "")
442 ;callerid= ; CallerID information for the endpoint (default: "")
443 ;callerid_privacy= ; Default privacy level (default: "")
444 ;callerid_tag= ; Internal id_tag for the endpoint (default: "")
445 ;context=default ; Dialplan context for inbound sessions (default:
447 ;direct_media_glare_mitigation=none ; Mitigation of direct media re INVITE
448 ; glare (default: "none")
449 ;direct_media_method=invite ; Direct Media method type (default: "invite")
450 ;connected_line_method=invite ; Connected line method type (default:
452 ;direct_media=yes ; Determines whether media may flow directly between
453 ; endpoints (default: "yes")
454 ;disable_direct_media_on_nat=no ; Disable direct media session refreshes when
455 ; NAT obstructs the media session (default:
457 ;disallow= ; Media Codec s to disallow (default: "")
458 ;dtmf_mode=rfc4733 ; DTMF mode (default: "rfc4733")
459 ;media_address= ; IP address used in SDP for media handling (default: "")
460 ;force_rport=yes ; Force use of return port (default: "yes")
461 ;ice_support=no ; Enable the ICE mechanism to help traverse NAT (default: "no")
462 ;identify_by=username ; Way s for Endpoint to be identified (default:
464 ;redirect_method=user ; How redirects received from an endpoint are handled
466 ;mailboxes= ; Mailbox es to be associated with (default: "")
467 ;moh_suggest=default ; Default Music On Hold class (default: "default")
468 ;outbound_auth= ; Authentication object used for outbound requests (default:
470 ;outbound_proxy= ; Proxy through which to send requests a full SIP URI
471 ; must be provided (default: "")
472 ;rewrite_contact=no ; Allow Contact header to be rewritten with the source
473 ; IP address port (default: "no")
474 ;rtp_ipv6=no ; Allow use of IPv6 for RTP traffic (default: "no")
475 ;rtp_symmetric=no ; Enforce that RTP must be symmetric (default: "no")
476 ;send_diversion=yes ; Send the Diversion header conveying the diversion
477 ; information to the called user agent (default: "yes")
478 ;send_pai=no ; Send the P Asserted Identity header (default: "no")
479 ;send_rpid=no ; Send the Remote Party ID header (default: "no")
480 ;timers_min_se=90 ; Minimum session timers expiration period (default:
482 ;timers=yes ; Session timers for SIP packets (default: "yes")
483 ;timers_sess_expires=1800 ; Maximum session timer expiration period
485 ;transport= ; Desired transport configuration (default: "")
486 ;trust_id_inbound=no ; Accept identification information received from this
487 ; endpoint (default: "no")
488 ;trust_id_outbound=no ; Send private identification details to the endpoint
490 ;type= ; Must be of type endpoint (default: "")
491 ;use_ptime=no ; Use Endpoint s requested packetisation interval (default:
493 ;use_avpf=no ; Determines whether res_pjsip will use and enforce usage of
494 ; AVPF for this endpoint (default: "no")
495 ;media_encryption=no ; Determines whether res_pjsip will use and enforce
496 ; usage of media encryption for this endpoint (default:
498 ;inband_progress=no ; Determines whether chan_pjsip will indicate ringing
499 ; using inband progress (default: "no")
500 ;call_group= ; The numeric pickup groups for a channel (default: "")
501 ;pickup_group= ; The numeric pickup groups that a channel can pickup (default:
503 ;named_call_group= ; The named pickup groups for a channel (default: "")
504 ;named_pickup_group= ; The named pickup groups that a channel can pickup
506 ;device_state_busy_at=0 ; The number of in use channels which will cause busy
507 ; to be returned as device state (default: "0")
508 ;t38_udptl=no ; Whether T 38 UDPTL support is enabled or not (default: "no")
509 ;t38_udptl_ec=none ; T 38 UDPTL error correction method (default: "none")
510 ;t38_udptl_maxdatagram=0 ; T 38 UDPTL maximum datagram size (default:
512 ;fax_detect=no ; Whether CNG tone detection is enabled (default: "no")
513 ;t38_udptl_nat=no ; Whether NAT support is enabled on UDPTL sessions
515 ;t38_udptl_ipv6=no ; Whether IPv6 is used for UDPTL Sessions (default:
517 ;tone_zone= ; Set which country s indications to use for channels created
518 ; for this endpoint (default: "")
519 ;language= ; Set the default language to use for channels created for this
520 ; endpoint (default: "")
521 ;one_touch_recording=no ; Determines whether one touch recording is allowed for
522 ; this endpoint (default: "no")
523 ;record_on_feature=automixmon ; The feature to enact when one touch recording
524 ; is turned on (default: "automixmon")
525 ;record_off_feature=automixmon ; The feature to enact when one touch recording
526 ; is turned off (default: "automixmon")
527 ;rtp_engine=asterisk ; Name of the RTP engine to use for channels created
528 ; for this endpoint (default: "asterisk")
529 ;allow_transfer=yes ; Determines whether SIP REFER transfers are allowed
530 ; for this endpoint (default: "yes")
531 ;sdp_owner=- ; String placed as the username portion of an SDP origin o line
533 ;sdp_session=Asterisk ; String used for the SDP session s line (default:
535 ;tos_audio=0 ; DSCP TOS bits for audio streams (default: "0")
536 ;tos_video=0 ; DSCP TOS bits for video streams (default: "0")
537 ;cos_audio=0 ; Priority for audio streams (default: "0")
538 ;cos_video=0 ; Priority for video streams (default: "0")
539 ;allow_subscribe=yes ; Determines if endpoint is allowed to initiate
540 ; subscriptions with Asterisk (default: "yes")
541 ;sub_min_expiry=0 ; The minimum allowed expiry time for subscriptions
542 ; initiated by the endpoint (default: "0")
543 ;from_user= ; Username to use in From header for requests to this endpoint
545 ;mwi_from_user= ; Username to use in From header for unsolicited MWI NOTIFYs to
546 ; this endpoint (default: "")
547 ;from_domain= ; Domain to user in From header for requests to this endpoint
549 ;dtls_verify= ; Verify that the provided peer certificate is valid (default:
551 ;dtls_rekey= ; Interval at which to renegotiate the TLS session and rekey
552 ; the SRTP session (default: "")
553 ;dtls_cert_file= ; Path to certificate file to present to peer (default:
555 ;dtls_private_key= ; Path to private key for certificate file (default:
557 ;dtls_cipher= ; Cipher to use for DTLS negotiation (default: "")
558 ;dtls_ca_file= ; Path to certificate authority certificate (default: "")
559 ;dtls_ca_path= ; Path to a directory containing certificate authority
560 ; certificates (default: "")
561 ;dtls_setup= ; Whether we are willing to accept connections connect to the
562 ; other party or both (default: "")
563 ;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
564 ; byte tags (default: "no")
565 ;set_var= ; Variable set on a channel involving the endpoint. For multiple
566 ; channel variables specify multiple 'set_var'(s)
568 ;==========================AUTH SECTION OPTIONS=========================
570 ; SYNOPSIS: Authentication type
571 ;auth_type=userpass ; Authentication type (default: "userpass")
572 ;nonce_lifetime=32 ; Lifetime of a nonce associated with this
573 ; authentication config (default: "32")
574 ;md5_cred= ; MD5 Hash used for authentication (default: "")
575 ;password= ; PlainText password used for authentication (default: "")
576 ;realm= ; SIP realm for endpoint (default: "")
577 ;type= ; Must be auth (default: "")
578 ;username= ; Username to use for account (default: "")
581 ;==========================DOMAIN_ALIAS SECTION OPTIONS=========================
583 ; SYNOPSIS: Domain Alias
584 ;type= ; Must be of type domain_alias (default: "")
585 ;domain= ; Domain to be aliased (default: "")
588 ;==========================TRANSPORT SECTION OPTIONS=========================
590 ; SYNOPSIS: SIP Transport
591 ;async_operations=1 ; Number of simultaneous Asynchronous Operations
593 ;bind= ; IP Address and optional port to bind to for this transport (default:
595 ;ca_list_file= ; File containing a list of certificates to read TLS ONLY
597 ;cert_file= ; Certificate file for endpoint TLS ONLY (default: "")
598 ;cipher= ; Preferred Cryptography Cipher TLS ONLY (default: "")
599 ;domain= ; Domain the transport comes from (default: "")
600 ;external_media_address= ; External IP address to use in RTP handling
602 ;external_signaling_address= ; External address for SIP signalling (default:
604 ;external_signaling_port=0 ; External port for SIP signalling (default:
606 ;method= ; Method of SSL transport TLS ONLY (default: "")
607 ;local_net= ; Network to consider local used for NAT purposes (default: "")
608 ;password= ; Password required for transport (default: "")
609 ;priv_key_file= ; Private key file TLS ONLY (default: "")
610 ;protocol=udp ; Protocol to use for SIP traffic (default: "udp")
611 ;require_client_cert= ; Require client certificate TLS ONLY (default: "")
612 ;type= ; Must be of type transport (default: "")
613 ;verify_client= ; Require verification of client certificate TLS ONLY (default:
615 ;verify_server= ; Require verification of server certificate TLS ONLY (default:
617 ;tos=0 ; Enable TOS for the signalling sent over this transport (default: "0")
618 ;cos=0 ; Enable COS for the signalling sent over this transport (default: "0")
621 ;==========================CONTACT SECTION OPTIONS=========================
623 ; SYNOPSIS: A way of creating an aliased name to a SIP URI
624 ;type= ; Must be of type contact (default: "")
625 ;uri= ; SIP URI to contact peer (default: "")
626 ;expiration_time= ; Time to keep alive a contact (default: "")
627 ;qualify_frequency=0 ; Interval at which to qualify a contact (default: "0")
628 ;outbound_proxy= ; Outbound proxy used when sending OPTIONS request
632 ;==========================AOR SECTION OPTIONS=========================
634 ; SYNOPSIS: The configuration for a location of an endpoint
635 ;contact= ; Permanent contacts assigned to AoR (default: "")
636 ;default_expiration=3600 ; Default expiration time in seconds for
637 ; contacts that are dynamically bound to an AoR
639 ;mailboxes= ; Mailbox es to be associated with (default: "")
640 ;maximum_expiration=7200 ; Maximum time to keep an AoR (default: "7200")
641 ;max_contacts=0 ; Maximum number of contacts that can bind to an AoR (default:
643 ;minimum_expiration=60 ; Minimum keep alive time for an AoR (default: "60")
644 ;remove_existing=no ; Determines whether new contacts replace existing ones
646 ;type= ; Must be of type aor (default: "")
647 ;qualify_frequency=0 ; Interval at which to qualify an AoR (default: "0")
648 ;authenticate_qualify=no ; Authenticates a qualify request if needed
650 ;outbound_proxy= ; Outbound proxy used when sending OPTIONS request
654 ;==========================SYSTEM SECTION OPTIONS=========================
656 ; SYNOPSIS: Options that apply to the SIP stack as well as other system-wide settings
657 ;timer_t1=500 ; Set transaction timer T1 value milliseconds (default: "500")
658 ;timer_b=32000 ; Set transaction timer B value milliseconds (default: "32000")
659 ;compact_headers=no ; Use the short forms of common SIP header names
661 ;threadpool_initial_size=0 ; Initial number of threads in the res_pjsip
662 ; threadpool (default: "0")
663 ;threadpool_auto_increment=5 ; The amount by which the number of threads is
664 ; incremented when necessary (default: "5")
665 ;threadpool_idle_timeout=60 ; Number of seconds before an idle thread
666 ; should be disposed of (default: "60")
667 ;threadpool_max_size=0 ; Maximum number of threads in the res_pjsip threadpool
668 ; A value of 0 indicates no maximum (default: "0")
669 ;type= ; Must be of type system (default: "")
671 ;==========================GLOBAL SECTION OPTIONS=========================
673 ; SYNOPSIS: Options that apply globally to all SIP communications
674 ;max_forwards=70 ; Value used in Max Forwards header for SIP requests
676 ;type= ; Must be of type global (default: "")
677 ;user_agent=Asterisk PBX SVN-branch-12-r404375 ; Value used in User Agent
678 ; header for SIP requests and
679 ; Server header for SIP
680 ; responses (default: "Asterisk
681 ; PBX SVN-branch-12-r404375")
682 ;default_outbound_endpoint=default_outbound_endpoint ; Endpoint to use when
683 ; sending an outbound
685 ; without a specified
686 ; endpoint (default: "d
687 ; efault_outbound_endpo
689 ;debug=no ; Enable/Disable SIP debug logging. Valid options include yes|no
690 ; or a host address (default: "no")
693 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl
694 ;==========================ACL SECTION OPTIONS=========================
696 ; SYNOPSIS: Access Control List
697 ;acl= ; List of IP ACL section names in acl conf (default: "")
698 ;contact_acl= ; List of Contact ACL section names in acl conf (default: "")
699 ;contact_deny= ; List of Contact header addresses to deny (default: "")
700 ;contact_permit= ; List of Contact header addresses to permit (default:
702 ;deny= ; List of IP addresses to deny access from (default: "")
703 ;permit= ; List of IP addresses to permit access from (default: "")
704 ;type= ; Must be of type acl (default: "")
709 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_registration
710 ;==========================REGISTRATION SECTION OPTIONS=========================
712 ; SYNOPSIS: The configuration for outbound registration
713 ;auth_rejection_permanent=yes ; Determines whether failed authentication
714 ; challenges are treated as permanent failures
716 ;client_uri= ; Client SIP URI used when attemping outbound registration
718 ;contact_user= ; Contact User to use in request (default: "")
719 ;expiration=3600 ; Expiration time for registrations in seconds
721 ;max_retries=10 ; Maximum number of registration attempts (default: "10")
722 ;outbound_auth= ; Authentication object to be used for outbound registrations
724 ;outbound_proxy= ; Outbound Proxy used to send registrations (default:
726 ;retry_interval=60 ; Interval in seconds between retries if outbound
727 ; registration is unsuccessful (default: "60")
728 ;forbidden_retry_interval=0 ; Interval used when receiving a 403 Forbidden
729 ; response (default: "0")
730 ;server_uri= ; SIP URI of the server to register against (default: "")
731 ;transport= ; Transport used for outbound authentication (default: "")
732 ;type= ; Must be of type registration (default: "")
737 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_endpoint_identifier_ip
738 ;==========================IDENTIFY SECTION OPTIONS=========================
740 ; SYNOPSIS: Identifies endpoints via source IP address
741 ;endpoint= ; Name of Endpoint (default: "")
742 ;match= ; IP addresses or networks to match against (default: "")
743 ;type= ; Must be of type identify (default: "")
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