[asterisk-commits] newtonr: branch 12 r404405 - /branches/12/configs/pjsip.conf.sample
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Dec 20 11:21:35 CST 2013
Author: newtonr
Date: Fri Dec 20 11:21:33 2013
New Revision: 404405
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=404405
Log:
Documentation: Updates for info about NAT-related settings and fixes for pjsip.conf.sample
Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity.
Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf.
I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options.
(issue ASTERISK-23004)
(closes issue ASTERISK-23004)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3086/
Modified:
branches/12/configs/pjsip.conf.sample
Modified: branches/12/configs/pjsip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/12/configs/pjsip.conf.sample?view=diff&rev=404405&r1=404404&r2=404405
==============================================================================
--- branches/12/configs/pjsip.conf.sample (original)
+++ branches/12/configs/pjsip.conf.sample Fri Dec 20 11:21:33 2013
@@ -81,7 +81,7 @@
;
; For the NAT transport example, be aware that the options starting with
; the prefix "external_" will only apply to communication with addresses
-; outside the range set with "localnet=".
+; outside the range set with "local_net=".
;
; IPv6: For endpoints using IPv6, remember to set "rtp_ipv6=yes" so that the RTP
; engine will also be able to bind to an IPv6 address.
@@ -102,7 +102,7 @@
;type=transport
;protocol=udp
;bind=0.0.0.0
-;localnet=192.0.2.0/24
+;local_net=192.0.2.0/24
;external_media_address=203.0.113.1
;external_signaling_address=203.0.113.1
@@ -197,7 +197,7 @@
;context=from-external
;disallow=all
;allow=ulaw
-;outbound_auth=mytrunk
+;outbound_auth=mytrunk_auth
;aors=mytrunk
; ;A few NAT relevant options that may come in handy.
;force_rport=yes ;It's a good idea to read the configuration help for each
@@ -291,13 +291,13 @@
;aggregate_mwi=yes
;mailboxes=6001 at default,7001 at default
-;mwifromuser=6001
+;mwi_from_user=6001
;
; Extension and Device state options
;
-;devicestate_busy_at=1
-;allowsubscribe=yes
-;subminexpiry=30
+;device_state_busy_at=1
+;allow_subscribe=yes
+;sub_min_expiry=30
;[6001]
;type=auth
@@ -310,6 +310,49 @@
;max_contacts=1
;contact=sip:6001 at 192.0.2.1:5060
+;===============ENDPOINT BEHIND NAT OR FIREWALL===============================
+;
+; This example assumes your transport is configured with a public IP and the
+; endpoint itself is behind NAT and maybe a firewall, rather than having
+; Asterisk behind NAT. For the sake of simplicity, we'll assume a typical
+; VOIP phone. The most important settings to configure are:
+;
+; * direct_media, to ensure Asterisk stays in the media path
+; * rtp_symmetric and force_rport options to help the far-end NAT/firewall
+;
+; Depending on the settings of your remote SIP device or NAT/firewall device
+; you may have to experiment with a combination of these settings.
+;
+; If both Asterisk and the remote phones are a behind NAT/firewall then you'll
+; have to make sure to use a transport with appropriate settings (as in the
+; transport-udp-nat example).
+;
+;[6002]
+;type=endpoint
+;transport=transport-udp
+;context=from-internal
+;disallow=all
+;allow=ulaw
+;auth=6002
+;aors=6002
+;direct_media=no
+;rtp_symmetric=yes
+;force_rport=yes
+;ice_support=yes ;This is specific to clients that support NAT traversal
+ ;for media via ICE,STUN,TURN. See the wiki at:
+ ;https://wiki.asterisk.org/wiki/x/D4FHAQ
+ ;for a deeper explanation of this topic.
+
+;[6002]
+;type=auth
+;auth_type=userpass
+;password=6002
+;username=6002
+
+;[6002]
+;type=aor
+;max_contacts=2
+
;============EXAMPLE ACL CONFIGURATION==========================================
;
@@ -330,7 +373,7 @@
;
;[acl]
;type=acl
-;contactacl=example_contact_acl1
+;contact_acl=example_contact_acl1
; Define your own ACL here in pjsip.conf and
; permit or deny by IP address or range.
@@ -346,10 +389,10 @@
;
;[acl]
;type=acl
-;contactdeny=0.0.0.0/0.0.0.0
-;contactpermit=209.16.236.0/24
-;contactpermit=209.16.236.1
-;contactpermit=209.16.236.2,209.16.236.3
+;contact_deny=0.0.0.0/0.0.0.0
+;contact_permit=209.16.236.0/24
+;contact_permit=209.16.236.1
+;contact_permit=209.16.236.2,209.16.236.3
; Restrict based on Contact Headers rather than IP and use
; advanced syntax. Note the bang symbol used for "NOT", so we can deny
@@ -357,8 +400,8 @@
;
;[acl]
;type=acl
-;contactdeny=0.0.0.0/0.0.0.0
-;contactpermit=209.16.236.0
+;contact_deny=0.0.0.0/0.0.0.0
+;contact_permit=209.16.236.0
;permit=209.16.236.0/24, !209.16.236.12/32
@@ -389,18 +432,20 @@
; NAT obstructs the media session (default:
; "no")
;disallow= ; Media Codec s to disallow (default: "")
-;dtmf_mode=rfc4733 ; DTMF mode (default: "rfc4733")
-;external_media_address= ; IP used for External Media handling (default:
- ; "")
+;dtmf_mode=rfc4733 ; DTMF mode (default: "rfc4733")
+;media_address= ; IP address used in SDP for media handling (default: "")
;force_rport=yes ; Force use of return port (default: "yes")
;ice_support=no ; Enable the ICE mechanism to help traverse NAT (default: "no")
;identify_by=username ; Way s for Endpoint to be identified (default:
; "username")
+;redirect_method=user ; How redirects received from an endpoint are handled
+ ; (default: "user")
;mailboxes= ; Mailbox es to be associated with (default: "")
-;moh_suggest=default ; Default Music On Hold class (default: "default")
+;moh_suggest=default ; Default Music On Hold class (default: "default")
;outbound_auth= ; Authentication object used for outbound requests (default:
; "")
-;outbound_proxy= ; Proxy through which to send requests (default: "")
+;outbound_proxy= ; Proxy through which to send requests a full SIP URI
+ ; must be provided (default: "")
;rewrite_contact=no ; Allow Contact header to be rewritten with the source
; IP address port (default: "no")
;rtp_ipv6=no ; Allow use of IPv6 for RTP traffic (default: "no")
@@ -429,66 +474,68 @@
; "no")
;inband_progress=no ; Determines whether chan_pjsip will indicate ringing
; using inband progress (default: "no")
-;call_group= ; The numeric pickup groups for a channel (default: "")
-;pickup_group= ; The numeric pickup groups that a channel can pickup (default:
- ; "")
-;named_call_group= ; The named pickup groups for a channel (default: "")
-;named_pickup_group= ; The named pickup groups that a channel can pickup
- ; (default: "")
-;device_state_busy_at=0 ; The number of in use channels which will cause busy
+;call_group= ; The numeric pickup groups for a channel (default: "")
+;pickup_group= ; The numeric pickup groups that a channel can pickup (default:
+ ; "")
+;named_call_group= ; The named pickup groups for a channel (default: "")
+;named_pickup_group= ; The named pickup groups that a channel can pickup
+ ; (default: "")
+;device_state_busy_at=0 ; The number of in use channels which will cause busy
; to be returned as device state (default: "0")
-;t38_udptl=no ; Whether T 38 UDPTL support is enabled or not (default: "no")
-;t38_udptl_ec=none ; T 38 UDPTL error correction method (default: "none")
-;t38_udptl_maxdatagram=0 ; T 38 UDPTL maximum datagram size (default: "0")
-;fax_detect=no ; Whether CNG tone detection is enabled (default: "no")
-;t38_udptl_nat=no ; Whether NAT support is enabled on UDPTL sessions
+;t38_udptl=no ; Whether T 38 UDPTL support is enabled or not (default: "no")
+;t38_udptl_ec=none ; T 38 UDPTL error correction method (default: "none")
+;t38_udptl_maxdatagram=0 ; T 38 UDPTL maximum datagram size (default:
+ ; "0")
+;fax_detect=no ; Whether CNG tone detection is enabled (default: "no")
+;t38_udptl_nat=no ; Whether NAT support is enabled on UDPTL sessions
; (default: "no")
-;t38_udptl_ipv6=no ; Whether IPv6 is used for UDPTL Sessions (default:
+;t38_udptl_ipv6=no ; Whether IPv6 is used for UDPTL Sessions (default:
; "no")
-;tone_zone= ; Set which country s indications to use for channels created
+;tone_zone= ; Set which country s indications to use for channels created
; for this endpoint (default: "")
;language= ; Set the default language to use for channels created for this
; endpoint (default: "")
;one_touch_recording=no ; Determines whether one touch recording is allowed for
; this endpoint (default: "no")
-;record_on_feature=automixmon ; The feature to enact when one touch recording
- ; is turned on (default: "automixmon")
-;record_off_feature=automixmon ; The feature to enact when one touch recording
- ; is turned off (default: "automixmon")
-;rtp_engine=asterisk ; Name of the RTP engine to use for channels created
- ; for this endpoint (default: "asterisk")
-;allow_transfer=yes ; Determines whether SIP REFER transfers are allowed
- ; for this endpoint (default: "yes")
-;sdp_owner=- ; String placed as the username portion of an SDP origin o line
- ; (default: "-")
-;sdp_session=Asterisk ; String used for the SDP session s line (default:
- ; "Asterisk")
+;record_on_feature=automixmon ; The feature to enact when one touch recording
+ ; is turned on (default: "automixmon")
+;record_off_feature=automixmon ; The feature to enact when one touch recording
+ ; is turned off (default: "automixmon")
+;rtp_engine=asterisk ; Name of the RTP engine to use for channels created
+ ; for this endpoint (default: "asterisk")
+;allow_transfer=yes ; Determines whether SIP REFER transfers are allowed
+ ; for this endpoint (default: "yes")
+;sdp_owner=- ; String placed as the username portion of an SDP origin o line
+ ; (default: "-")
+;sdp_session=Asterisk ; String used for the SDP session s line (default:
+ ; "Asterisk")
;tos_audio=0 ; DSCP TOS bits for audio streams (default: "0")
;tos_video=0 ; DSCP TOS bits for video streams (default: "0")
;cos_audio=0 ; Priority for audio streams (default: "0")
;cos_video=0 ; Priority for video streams (default: "0")
-;allow_subscribe=yes ; Determines if endpoint is allowed to initiate
- ; subscriptions with Asterisk (default: "yes")
-;sub_min_expiry=0 ; The minimum allowed expiry time for subscriptions initiated
- ; by the endpoint (default: "0")
-;from_user= ; Username to use in From header for requests to this endpoint
- ; (default: "")
-;mwi_from_user= ; Username to use in From header for unsolicited MWI NOTIFYs to
- ; this endpoint (default: "")
-;fromdomain= ; Domain to user in From header for requests to this endpoint
+;allow_subscribe=yes ; Determines if endpoint is allowed to initiate
+ ; subscriptions with Asterisk (default: "yes")
+;sub_min_expiry=0 ; The minimum allowed expiry time for subscriptions
+ ; initiated by the endpoint (default: "0")
+;from_user= ; Username to use in From header for requests to this endpoint
; (default: "")
-;dtls_verify= ; Verify that the provided peer certificate is valid (default:
- ; "")
-;dtls_rekey= ; Interval at which to renegotiate the TLS session and rekey
- ; the SRTP session (default: "")
-;dtls_cert_file= ; Path to certificate file to present to peer (default: "")
-;dtls_private_key= ; Path to private key for certificate file (default:
+;mwi_from_user= ; Username to use in From header for unsolicited MWI NOTIFYs to
+ ; this endpoint (default: "")
+;from_domain= ; Domain to user in From header for requests to this endpoint
+ ; (default: "")
+;dtls_verify= ; Verify that the provided peer certificate is valid (default:
+ ; "")
+;dtls_rekey= ; Interval at which to renegotiate the TLS session and rekey
+ ; the SRTP session (default: "")
+;dtls_cert_file= ; Path to certificate file to present to peer (default:
; "")
-;dtls_cipher= ; Cipher to use for DTLS negotiation (default: "")
-;dtls_ca_file= ; Path to certificate authority certificate (default: "")
-;dtls_ca_path= ; Path to a directory containing certificate authority
- ; certificates (default: "")
-;dtls_setup= ; Whether we are willing to accept connections connect to the
+;dtls_private_key= ; Path to private key for certificate file (default:
+ ; "")
+;dtls_cipher= ; Cipher to use for DTLS negotiation (default: "")
+;dtls_ca_file= ; Path to certificate authority certificate (default: "")
+;dtls_ca_path= ; Path to a directory containing certificate authority
+ ; certificates (default: "")
+;dtls_setup= ; Whether we are willing to accept connections connect to the
; other party or both (default: "")
;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
; byte tags (default: "no")
@@ -502,7 +549,7 @@
; authentication config (default: "32")
;md5_cred= ; MD5 Hash used for authentication (default: "")
;password= ; PlainText password used for authentication (default: "")
-;realm=asterisk ; SIP realm for endpoint (default: "asterisk")
+;realm= ; SIP realm for endpoint (default: "")
;type= ; Must be auth (default: "")
;username= ; Username to use for account (default: "")
@@ -526,16 +573,16 @@
;cert_file= ; Certificate file for endpoint TLS ONLY (default: "")
;cipher= ; Preferred Cryptography Cipher TLS ONLY (default: "")
;domain= ; Domain the transport comes from (default: "")
-;external_media_address= ; External Address to use in RTP handling
+;external_media_address= ; External IP address to use in RTP handling
; (default: "")
;external_signaling_address= ; External address for SIP signalling (default:
; "")
;external_signaling_port=0 ; External port for SIP signalling (default:
; "0")
;method= ; Method of SSL transport TLS ONLY (default: "")
-;local_net= ; Network to consider local used for NAT purposes (default: "")
+;local_net= ; Network to consider local used for NAT purposes (default: "")
;password= ; Password required for transport (default: "")
-;priv_key_file= ; Private key file TLS ONLY (default: "")
+;priv_key_file= ; Private key file TLS ONLY (default: "")
;protocol=udp ; Protocol to use for SIP traffic (default: "udp")
;require_client_cert= ; Require client certificate TLS ONLY (default: "")
;type= ; Must be of type transport (default: "")
@@ -554,6 +601,8 @@
;uri= ; SIP URI to contact peer (default: "")
;expiration_time= ; Time to keep alive a contact (default: "")
;qualify_frequency=0 ; Interval at which to qualify a contact (default: "0")
+;outbound_proxy= ; Outbound proxy used when sending OPTIONS request
+ ; (default: "")
;==========================AOR SECTION OPTIONS=========================
@@ -574,14 +623,16 @@
;qualify_frequency=0 ; Interval at which to qualify an AoR (default: "0")
;authenticate_qualify=no ; Authenticates a qualify request if needed
; (default: "no")
+;outbound_proxy= ; Outbound proxy used when sending OPTIONS request
+ ; (default: "")
;==========================SYSTEM SECTION OPTIONS=========================
;[system]
; SYNOPSIS: Options that apply to the SIP stack as well as other system-wide settings
-;timer_t1=500 ; Set transaction timer T1 value milliseconds (default: "500")
-;timer_b=32000 ; Set transaction timer B value milliseconds (default: "32000")
-;compact_headers=no ; Use the short forms of common SIP header names
+;timer_t1=500 ; Set transaction timer T1 value milliseconds (default: "500")
+;timer_b=32000 ; Set transaction timer B value milliseconds (default: "32000")
+;compact_headers=no ; Use the short forms of common SIP header names
; (default: "no")
;threadpool_initial_size=0 ; Initial number of threads in the res_pjsip
; threadpool (default: "0")
@@ -597,28 +648,37 @@
;==========================GLOBAL SECTION OPTIONS=========================
;[global]
; SYNOPSIS: Options that apply globally to all SIP communications
-;max_forwards=70 ; Value used in Max Forwards header for SIP requests (default:
- ; "70")
+;max_forwards=70 ; Value used in Max Forwards header for SIP requests
+ ; (default: "70")
;type= ; Must be of type global (default: "")
-;user_agent= ; Value used in User Agent header for SIP requests and Server
- ; header for SIP responses (default: Populated by Asterisk
- ; Version)
-;default_outbound_endpoint= ; Endpoint to use when sending an outbound request
- ; to a URI without a specified endpoint.
- ; (default: "default_outbound_endpoint")
+;user_agent=Asterisk PBX SVN-branch-12-r404375 ; Value used in User Agent
+ ; header for SIP requests and
+ ; Server header for SIP
+ ; responses (default: "Asterisk
+ ; PBX SVN-branch-12-r404375")
+;default_outbound_endpoint=default_outbound_endpoint ; Endpoint to use when
+ ; sending an outbound
+ ; request to a URI
+ ; without a specified
+ ; endpoint (default: "d
+ ; efault_outbound_endpo
+ ; int")
+
+
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl
;==========================ACL SECTION OPTIONS=========================
;[acl]
; SYNOPSIS: Access Control List
-;acl= ; Name of IP ACL (default: "")
-;contact_acl= ; Name of Contact ACL (default: "")
-;contact_deny= ; List of Contact Header addresses to Deny (default: "")
-;contact_permit= ; List of Contact Header addresses to Permit (default: "")
-;deny= ; List of IP domains to deny access from (default: "")
-;permit= ; List of IP domains to allow access from (default: "")
-;type= ; Must be of type security (default: "")
+;acl= ; List of IP ACL section names in acl conf (default: "")
+;contact_acl= ; List of Contact ACL section names in acl conf (default: "")
+;contact_deny= ; List of Contact header addresses to deny (default: "")
+;contact_permit= ; List of Contact header addresses to permit (default:
+ ; "")
+;deny= ; List of IP addresses to deny access from (default: "")
+;permit= ; List of IP addresses to permit access from (default: "")
+;type= ; Must be of type acl (default: "")
@@ -642,6 +702,8 @@
; "")
;retry_interval=60 ; Interval in seconds between retries if outbound
; registration is unsuccessful (default: "60")
+;forbidden_retry_interval=0 ; Interval used when receiving a 403 Forbidden
+ ; response (default: "0")
;server_uri= ; SIP URI of the server to register against (default: "")
;transport= ; Transport used for outbound authentication (default: "")
;type= ; Must be of type registration (default: "")
@@ -652,11 +714,7 @@
; MODULE PROVIDING BELOW SECTION(S): res_pjsip_endpoint_identifier_ip
;==========================IDENTIFY SECTION OPTIONS=========================
;[identify]
-; SYNOPSIS: NEEDS A SYNOPSIS
+; SYNOPSIS: Identifies endpoints via source IP address
;endpoint= ; Name of Endpoint (default: "")
;match= ; IP addresses or networks to match against (default: "")
;type= ; Must be of type identify (default: "")
-
-
-
-
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