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About

mod_sofia is the SIP endpoint implemented by FreeSWITCH.

Configuration

sofia.conf.xml

accept-blind-auth|true,false
accept-blind-reg|true,false
aggressive-nat-detection|true,false alias|arbitrary all-reg-options-ping|true,false apply-candidate-acl|acl apply-inbound-acl|acl apply-nat-acl|acl apply-proxy-acl|acl apply-register-acl|acl auth-all-packets|true,false auth-calls|true,false auth-messages|true,false auto-jitterbuffer-msec|msc greater than 19 auto-restart|true,false auto-rtp-bugs|clear,CISCO_SKIP_MARK_BIT_2833,SONUS_SEND_INVALID_TIMESTAMP_2833,IGNORE_MARK_BIT,SEND_LINEAR_TIMESTAMPS,START_SEQ_AT_ZERO,NEVER_SEND_MARKER,IGNORE_DTMF_DURATION,ACCEPT_ANY_PACKETS,GEN_ONE_GEN_ALL,CHANGE_SSRC_ON_MARKER,FLUSH_JB_ON_DTMF bind-params|transport=tcp,udp,tls bitpacking|aal2 (defaults to 3550 packing) caller-id-type|pid,rpid,none (default is none) capture-server|udp:x.x.x.x:port challenge-realm|arbitrary cid-in-1xx|true,false cng-pt|payload number codec-prefs|codec list comma sep. confirm-blind-transfer|true,false contact-user|arbitrary context|context to process calls in. db-inner-post-trans-execute|? db-inner-pre-trans-execute|? dbname|arbitrary db-post-trans-execute|? db-pre-trans-execute|? debug-presence]|number debug-sla|number dialplan|which dialplan to process calls with (XML) disable-hold|true,false disable-naptr|true,false disable-register|true,false disable-rtp-auto-adjust|true,false disable-srv|true,false disable-srv503|true,false disable-transcoding|true,false disable-transfer|true,false dtmf-duration| SWITCH_DEFAULT_DTMF_DURATION which is 2000 dtmf-type|info,rf2833,none enable-100rel|true,false enable-3pcc|true,false enable-compact-headers|true,false enable-rfc-5626|true,false enable-soa|true,false enable-timer|true,false extended-info-parsing|true,false ext-rtp-ip|ipaddress ext-sip-ip|ipaddress ext-sip-port|port fire-message-events|true,false force-publish-expires|number force-register-db-domain|domain force-register-domain|domain force-subscription-domain|number force-subscription-expires|number forward-unsolicited-mwi-notify|true,false full-id-in-dialplan|true,false hold-music|arbitrary ignore-183nosdp|true,false inbound-bypass-media|true,false inbound-codec-negotiation|greedy,scrooge (default both) inbound-codec-prefs|codec list inbound-late-negotiation|true,false inbound-no-media|true,false inbound-proxy-media|true,false inbound-reg-force-matching-username|true,false inbound-reg-in-new-thread|true,false inbound-use-callid-as-uuid|true,false inbound-zrtp-passthru|true,false in-dialog-chat|true,false liberal-dtmf|true,false local-network-acl|acl log-auth-failures|true,false log-level|number manage-presence|passive,pnp,true manage-shared-appearance|true,false manual-redirect|true,false manual-rtp-bugs|rtp_bug_list manual-video-rtp-bugs|rtp_bug_list max-calls|number max-proceeding|number max-registrations-per-extension|number max-reg-threads|number media-option|resume-media-on-hold,bypass-media-after-att-xfer,bypass-media-after-hold minimum-session-expires|number multiple-registrations|number mwi-use-reg-callid|true,false nat-options-ping|udp-only,true (default off) NDLB-allow-bad-iananame|true,false NDLB-allow-crypto-in-avp|true,false NDLB-allow-nondup-sdp|true,false NDLB-broken-auth-hash|true,false NDLB-expires-in-register-response|true,false NDLB-force-rport|safe,disabled,client-only,client-only NDLB-received-in-nat-reg-contact|true,false NDLB-sendrecv-in-session|true,false NDLB-support-asterisk-missing-srtp-auth|true,false nonce-ttl|number odbc-dsn|odbc dsn info outbound-codec-prefs|codec list outbound-use-uuid-as-callid|true,false parse-all-invite-headers|true,false pass-callee-id|true,false p-asserted-id-parse|true,false pass-rfc2833|true,false pnp-provision-url|arbitrary presence-disable-early|true,false presence-hold-state|confirmed,terminated presence-hosts|hostnames presence-privacy|true,false presence-probe-on-register|true,false presence-proto-lookup|true profile-standby|true,false record-path|path record-template|template to expand reg-deny-binding-fetch-and-no-lookup|true,false registration-thread-frequency|number default IREG_SECONDS 30 renegotiate-codec-on-hold|true,false renegotiate-codec-on-reinvite|true,false require-secure-rtp|true,false reuse-connections|true,false rfc2833-pt|pt rtcp-audio-interval-msec|msec rtcp-video-interval-msec|msec rtp-autofix-timing|true,false rtp-digit-delay|number greater than 0 rtp-hold-timeout-sec|seconds rtp-ip|ip rtp-notimer-during-bridge|true,false rtp-rewrite-timestamps|true,false rtp-timeout-sec|seconds rtp-timer-name|string send-display-update|true,false send-message-query-on-register|true,false send-presence-on-register|true,false session-timeout|seconds shutdown-on-fail|true,false sip-capture|udp:ip:port sip-domain|domain sip-expires-late-margin|seconds sip-expires-max-deviation|seconds sip-force-expires|seconds sip-ip|ip sip-messages-respond-200-ok|true,false sip-options-respond-503-on-busy|true,false sip-port|port sip-subscription-max-deviation|seconds sip-trace|true,false socket-tcp-keepalive|true,false suppress-cng|true,false t38-passthru|true,false tcp-keepalive|milliseconds tcp-ping2pong|milliseconds tcp-pingpong|milliseconds timer-T1|default 500ms (sofia defaults it to 1000 to prevent race) timer-T1X64|default 32000 timer-T2|default 4000 timer-T4|4000 tls-bind-params|transport=tls or %s;transport=tls tls-cert-dir|dir tls-ciphers|ciper list tls-only|true,false tls-passphrase|arbitrary tls-sip-port|port tls-timeout|greater than 0 or 300 tls-verify-date|number tls-verify-depth|number tls-verify-in-subjects|? tls-verify-policy|? tls-version|sslv2,sslv3,sslv23,tlsv1,tlsv1.1,tlsv1.2 tracelevel|number track-calls|true,false unregister-on-options-fail|true,false user-agent-filter|regex user-agent-string|string username|sdp username vad|true,false watchdog-enabled|true,false watchdog-event-timeout|seconds watchdog-step-timeout|seconds ws-binding|ip:port or :port wss-binding|ip:port or :port

 



Call ID

inbound-use-callid-as-uuid

On inbound calls make the uuid of the session equal to the SIP call id of that call.

<param name="inbound-use-callid-as-uuid" value="true"/>

Internal sip_profile configuration.

 


 

See Also

Sofia SIP Stack

 

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