Oliver Smith | 667f19b | 2019-11-08 18:16:30 +0100 | [diff] [blame] | 1 | <profile name="internal"> |
| 2 | <!-- |
| 3 | This is a sofia sip profile/user agent. This will service exactly one ip and port. |
| 4 | In FreeSWITCH you can run multiple sip user agents on their own ip and port. |
| 5 | |
| 6 | When you hear someone say "sofia profile" this is what they are talking about. |
| 7 | --> |
| 8 | |
| 9 | <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> |
| 10 | <!--aliases are other names that will work as a valid profile name for this profile--> |
| 11 | <aliases> |
| 12 | <!-- |
| 13 | <alias name="default"/> |
| 14 | --> |
| 15 | </aliases> |
| 16 | <!-- Outbound Registrations --> |
| 17 | |
| 18 | <domains> |
| 19 | <!-- indicator to parse the directory for domains with parse="true" to get gateways--> |
| 20 | <!--<domain name="$${domain}" parse="true"/>--> |
| 21 | <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile --> |
| 22 | <!--<domain name="all" alias="true" parse="true"/>--> |
| 23 | <domain name="all" alias="true" parse="false"/> |
| 24 | </domains> |
| 25 | |
| 26 | <settings> |
| 27 | |
| 28 | |
| 29 | <!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var --> |
| 30 | <!-- <param name="rtp-digit-delay" value="40"/>--> |
| 31 | |
| 32 | <!-- |
| 33 | When calls are in no media this will bring them back to media |
| 34 | when you press the hold button. |
| 35 | --> |
| 36 | <!--<param name="media-option" value="resume-media-on-hold"/> --> |
| 37 | <!-- |
| 38 | This will allow a call after an attended transfer go back to |
| 39 | bypass media after an attended transfer. |
| 40 | --> |
| 41 | <!--<param name="media-option" value="bypass-media-after-att-xfer"/>--> |
| 42 | <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> --> |
| 43 | <param name="debug" value="0"/> |
| 44 | <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. --> |
| 45 | <!-- <param name="shutdown-on-fail" value="true"/> --> |
| 46 | <param name="sip-trace" value="no"/> |
| 47 | <param name="sip-capture" value="no"/> |
| 48 | |
| 49 | <!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing --> |
| 50 | <!-- <param name="presence-proto-lookup" value="true"/> --> |
| 51 | |
| 52 | |
| 53 | <!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO --> |
| 54 | <param name="liberal-dtmf" value="true"/> |
| 55 | |
| 56 | |
| 57 | <!-- |
| 58 | Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop |
| 59 | responding. These options allow you to enable and control a watchdog |
| 60 | on the Sofia SIP stack so that if it stops responding for the |
| 61 | specified number of milliseconds, it will cause FreeSWITCH to crash |
| 62 | immediately. This is useful if you run in an HA environment and |
| 63 | need to ensure automated recovery from such a condition. Note that if |
| 64 | your server is idle a lot, the watchdog may fire due to not receiving |
| 65 | any SIP messages. Thus, if you expect your system to be idle, you |
| 66 | should leave the watchdog disabled. It can be toggled on and off |
| 67 | through the FreeSWITCH CLI either on an individual profile basis or |
| 68 | globally for all profiles. So, if you run in an HA environment with a |
| 69 | master and slave, you should use the CLI to make sure the watchdog is |
| 70 | only enabled on the master. |
| 71 | If such crash occurs, FreeSWITCH will dump core if allowed. The |
| 72 | stacktrace will include function watchdog_triggered_abort(). |
| 73 | --> |
| 74 | <param name="watchdog-enabled" value="no"/> |
| 75 | <param name="watchdog-step-timeout" value="30000"/> |
| 76 | <param name="watchdog-event-timeout" value="30000"/> |
| 77 | |
| 78 | <param name="log-auth-failures" value="false"/> |
| 79 | <param name="forward-unsolicited-mwi-notify" value="false"/> |
| 80 | |
| 81 | <param name="context" value="public"/> |
| 82 | <param name="rfc2833-pt" value="101"/> |
| 83 | <!-- port to bind to for sip traffic --> |
Neels Hofmeyr | 96a12a1 | 2019-12-04 03:43:12 +0100 | [diff] [blame] | 84 | <param name="sip-port" value="${PBX_SIP_PORT}"/> |
Oliver Smith | 667f19b | 2019-11-08 18:16:30 +0100 | [diff] [blame] | 85 | <param name="dialplan" value="XML"/> |
| 86 | <param name="dtmf-duration" value="2000"/> |
| 87 | <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> |
| 88 | <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> |
| 89 | <param name="rtp-timer-name" value="soft"/> |
| 90 | <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES --> |
| 91 | <param name="rtp-ip" value="$${local_ip_v4}"/> |
| 92 | <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES --> |
Neels Hofmeyr | 96a12a1 | 2019-12-04 03:43:12 +0100 | [diff] [blame] | 93 | <param name="sip-ip" value="${PBX_SIP}"/> |
Oliver Smith | 667f19b | 2019-11-08 18:16:30 +0100 | [diff] [blame] | 94 | <!--<param name="hold-music" value="$${hold_music}"/>--> |
| 95 | <param name="apply-nat-acl" value="nat.auto"/> |
| 96 | |
| 97 | |
| 98 | <!-- (default true) set to false if you do not wish to have called party info in 1XX responses --> |
| 99 | <!-- <param name="cid-in-1xx" value="false"/> --> |
| 100 | |
| 101 | <!-- extended info parsing --> |
| 102 | <!-- <param name="extended-info-parsing" value="true"/> --> |
| 103 | |
| 104 | <param name="aggressive-nat-detection" value="false"/> |
| 105 | <!-- |
| 106 | There are known issues (asserts and segfaults) when 100rel is enabled. |
| 107 | It is not recommended to enable 100rel at this time. |
| 108 | --> |
| 109 | <!--<param name="enable-100rel" value="true"/>--> |
| 110 | |
| 111 | <!-- uncomment if you don't wish to try a next SRV destination on 503 response --> |
| 112 | <!-- RFC3263 Section 4.3 --> |
| 113 | <!--<param name="disable-srv503" value="true"/>--> |
| 114 | |
| 115 | <!-- Enable Compact SIP headers. --> |
| 116 | <!--<param name="enable-compact-headers" value="true"/>--> |
| 117 | <!-- |
| 118 | enable/disable session timers |
| 119 | --> |
| 120 | <!--<param name="enable-timer" value="false"/>--> |
| 121 | <!--<param name="minimum-session-expires" value="120"/>--> |
| 122 | <param name="apply-inbound-acl" value="domains"/> |
| 123 | <!-- |
| 124 | This defines your local network, by default we detect your local network |
| 125 | and create this localnet.auto ACL for this. |
| 126 | --> |
| 127 | <param name="local-network-acl" value="localnet.auto"/> |
| 128 | <param name="apply-register-acl" value="domains"/> |
| 129 | <param name="apply-candidate-acl" value="domains"/> |
| 130 | <!--<param name="dtmf-type" value="info"/>--> |
| 131 | |
| 132 | |
| 133 | <!-- 'true' means every time 'first-only' means on the first register --> |
| 134 | <!--<param name="send-message-query-on-register" value="true"/>--> |
| 135 | |
| 136 | <!-- 'true' means every time 'first-only' means on the first register --> |
| 137 | <!--<param name="send-presence-on-register" value="first-only"/> --> |
| 138 | |
| 139 | |
| 140 | <!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable --> |
| 141 | <!-- Remote-Party-ID header --> |
| 142 | <!--<param name="caller-id-type" value="rpid"/>--> |
| 143 | |
| 144 | <!-- P-*-Identity family of headers --> |
| 145 | <!--<param name="caller-id-type" value="pid"/>--> |
| 146 | |
| 147 | <!-- neither one --> |
| 148 | <!--<param name="caller-id-type" value="none"/>--> |
| 149 | |
| 150 | |
| 151 | |
| 152 | <param name="record-path" value="$${recordings_dir}"/> |
| 153 | <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> |
| 154 | <!--enable to use presence --> |
| 155 | <param name="manage-presence" value="false"/> |
| 156 | <!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info --> |
| 157 | <!--<param name="presence-probe-on-register" value="true"/>--> |
| 158 | <!--<param name="manage-shared-appearance" value="true"/>--> |
| 159 | <!-- used to share presence info across sofia profiles --> |
| 160 | <!-- Name of the db to use for this profile --> |
| 161 | <!--<param name="dbname" value="share_presence"/>--> |
| 162 | <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/> |
| 163 | <param name="presence-privacy" value="$${presence_privacy}"/> |
| 164 | <!-- ************************************************* --> |
| 165 | |
| 166 | <!-- This setting is for AAL2 bitpacking on G726 --> |
| 167 | <!-- <param name="bitpacking" value="aal2"/> --> |
| 168 | <!--max number of open dialogs in proceeding --> |
| 169 | <!--<param name="max-proceeding" value="1000"/>--> |
| 170 | <!--session timers for all call to expire after the specified seconds --> |
| 171 | <!--<param name="session-timeout" value="1800"/>--> |
| 172 | <!-- Can be 'true' or 'contact' --> |
| 173 | <!--<param name="multiple-registrations" value="contact"/>--> |
| 174 | <!--set to 'greedy' if you want your codec list to take precedence --> |
| 175 | <param name="inbound-codec-negotiation" value="generous"/> |
| 176 | <!-- if you want to send any special bind params of your own --> |
| 177 | <!--<param name="bind-params" value="transport=udp"/>--> |
| 178 | <!--<param name="unregister-on-options-fail" value="true"/>--> |
| 179 | <!-- Send an OPTIONS packet to all registered endpoints --> |
| 180 | <!--<param name="all-reg-options-ping" value="true"/>--> |
| 181 | <!-- Send an OPTIONS packet to NATed registered endpoints. Can be 'true' or 'udp-only'. --> |
| 182 | <!--<param name="nat-options-ping" value="true"/>--> |
| 183 | |
| 184 | <!-- TLS: disabled by default, set to "true" to enable --> |
| 185 | <param name="tls" value="$${internal_ssl_enable}"/> |
| 186 | <!-- Set to true to not bind on the normal sip-port but only on the TLS port --> |
| 187 | <param name="tls-only" value="false"/> |
| 188 | <!-- additional bind parameters for TLS --> |
| 189 | <param name="tls-bind-params" value="transport=tls"/> |
| 190 | <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) --> |
| 191 | <param name="tls-sip-port" value="$${internal_tls_port}"/> |
| 192 | <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) --> |
| 193 | <!--<param name="tls-cert-dir" value=""/>--> |
| 194 | <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files --> |
| 195 | <param name="tls-passphrase" value=""/> |
| 196 | <!-- Verify the date on TLS certificates --> |
| 197 | <param name="tls-verify-date" value="true"/> |
| 198 | <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate --> |
| 199 | <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe --> |
| 200 | <param name="tls-verify-policy" value="none"/> |
| 201 | <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none --> |
| 202 | <param name="tls-verify-depth" value="2"/> |
| 203 | <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe --> |
| 204 | <param name="tls-verify-in-subjects" value=""/> |
| 205 | <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 --> |
| 206 | <param name="tls-version" value="$${sip_tls_version}"/> |
| 207 | |
| 208 | <!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data) |
| 209 | (reduces delay on latent connections default true, must be disabled explicitly)--> |
| 210 | <!--<param name="rtp-autoflush-during-bridge" value="false"/>--> |
| 211 | |
| 212 | <!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)--> |
| 213 | <!--<param name="rtp-rewrite-timestamps" value="true"/>--> |
| 214 | <!--<param name="pass-rfc2833" value="true"/>--> |
| 215 | <!--If you have ODBC support and a working dsn you can use it instead of SQLite--> |
| 216 | <!--<param name="odbc-dsn" value="dsn:user:pass"/>--> |
| 217 | |
| 218 | <!-- Or, if you have PGSQL support, you can use that --> |
| 219 | <!--<param name="odbc-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'" />--> |
| 220 | |
| 221 | <!--Uncomment to set all inbound calls to no media mode--> |
| 222 | <!--<param name="inbound-bypass-media" value="true"/>--> |
| 223 | |
| 224 | <!--Uncomment to set all inbound calls to proxy media mode--> |
| 225 | <!--<param name="inbound-proxy-media" value="true"/>--> |
| 226 | |
| 227 | <!-- Let calls hit the dialplan before selecting codec for the a-leg --> |
| 228 | <param name="inbound-late-negotiation" value="false"/> |
| 229 | |
| 230 | <!-- Allow ZRTP clients to negotiate end-to-end security associations (also enables late negotiation) --> |
| 231 | <param name="inbound-zrtp-passthru" value="true"/> |
| 232 | |
| 233 | <!-- this lets anything register --> |
| 234 | <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> |
| 235 | <!-- <param name="accept-blind-reg" value="true"/> --> |
| 236 | |
| 237 | <!-- accept any authentication without actually checking (not a good feature for most people) --> |
| 238 | <!-- <param name="accept-blind-auth" value="true"/> --> |
| 239 | |
| 240 | <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable --> |
| 241 | <!-- <param name="suppress-cng" value="true"/> --> |
| 242 | |
| 243 | <!--TTL for nonce in sip auth--> |
| 244 | <param name="nonce-ttl" value="60"/> |
| 245 | <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec |
| 246 | that the originator is using--> |
| 247 | <!--<param name="disable-transcoding" value="true"/>--> |
| 248 | <!-- Handle 302 Redirect in the dialplan --> |
| 249 | <!--<param name="manual-redirect" value="true"/> --> |
| 250 | <!-- Disable Transfer --> |
| 251 | <!--<param name="disable-transfer" value="true"/> --> |
| 252 | <!-- Disable Register --> |
| 253 | <!--<param name="disable-register" value="true"/> --> |
| 254 | <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash --> |
| 255 | <!--<param name="NDLB-broken-auth-hash" value="true"/>--> |
| 256 | <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling --> |
| 257 | <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>--> |
| 258 | <param name="auth-calls" value="$${internal_auth_calls}"/> |
| 259 | <!-- Force the user and auth-user to match. --> |
| 260 | <param name="inbound-reg-force-matching-username" value="true"/> |
| 261 | <!-- on authed calls, authenticate *all* the packets not just invite --> |
| 262 | <param name="auth-all-packets" value="false"/> |
| 263 | |
| 264 | <!-- external_sip_ip |
| 265 | Used as the public IP address for SDP. |
| 266 | Can be an one of: |
| 267 | ip address - "12.34.56.78" |
| 268 | a stun server lookup - "stun:stun.server.com" |
| 269 | a DNS name - "host:host.server.com" |
| 270 | auto - Use guessed ip. |
| 271 | auto-nat - Use ip learned from NAT-PMP or UPNP |
| 272 | --> |
| 273 | <param name="ext-rtp-ip" value="$${local_ip_v4}"/> |
| 274 | <param name="ext-sip-ip" value="$${local_ip_v4}"/> |
| 275 | |
| 276 | <!-- rtp inactivity timeout --> |
| 277 | <param name="rtp-timeout-sec" value="300"/> |
| 278 | <param name="rtp-hold-timeout-sec" value="1800"/> |
| 279 | <!-- VAD choose one (out is a good choice); --> |
| 280 | <!-- <param name="vad" value="in"/> --> |
| 281 | <!-- <param name="vad" value="out"/> --> |
| 282 | <!-- <param name="vad" value="both"/> --> |
| 283 | <!--<param name="alias" value="sip:10.0.1.251:5555"/>--> |
| 284 | <!-- |
| 285 | These are enabled to make the default config work better out of the box. |
| 286 | If you need more than ONE domain you'll need to not use these options. |
| 287 | |
| 288 | --> |
| 289 | <!--all inbound reg will look in this domain for the users --> |
| 290 | <param name="force-register-domain" value="$${domain}"/> |
| 291 | <!--force the domain in subscriptions to this value --> |
| 292 | <param name="force-subscription-domain" value="$${domain}"/> |
| 293 | <!--all inbound reg will stored in the db using this domain --> |
| 294 | <param name="force-register-db-domain" value="$${domain}"/> |
| 295 | |
| 296 | |
| 297 | <!-- uncomment for sip over websocket support --> |
| 298 | <!--<param name="ws-binding" value=":5066"/>--> |
| 299 | |
| 300 | <!-- uncomment for sip over secure websocket support --> |
| 301 | <!-- You need wss.pem in /usr/local/freeswitch/certs for wss --> |
| 302 | <!--<param name="wss-binding" value=":7443"/>--> |
| 303 | |
| 304 | |
| 305 | <!--<param name="delete-subs-on-register" value="false"/>--> |
| 306 | |
| 307 | <!-- launch a new thread to process each new inbound register when using heavier backends --> |
| 308 | <!-- <param name="inbound-reg-in-new-thread" value="true"/> --> |
| 309 | |
| 310 | <!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call--> |
| 311 | <!--<param name="rtcp-audio-interval-msec" value="5000"/>--> |
| 312 | <!--<param name="rtcp-video-interval-msec" value="5000"/>--> |
| 313 | |
| 314 | <!--force suscription expires to a lower value than requested--> |
| 315 | <!--<param name="force-subscription-expires" value="60"/>--> |
| 316 | |
| 317 | <!-- add a random deviation to the expires value of the 202 Accepted --> |
| 318 | <!--<param name="sip-subscription-max-deviation" value="120"/>--> |
| 319 | |
| 320 | <!-- disable register and transfer which may be undesirable in a public switch --> |
| 321 | <!--<param name="disable-transfer" value="true"/>--> |
| 322 | <!--<param name="disable-register" value="true"/>--> |
| 323 | |
| 324 | <!-- |
| 325 | enable-3pcc can be set to either 'true' or 'proxy', true accepts the call |
| 326 | right away, proxy waits until the call has been answered then sends accepts |
| 327 | --> |
| 328 | <!--<param name="enable-3pcc" value="true"/>--> |
| 329 | |
| 330 | <!-- use at your own risk or if you know what this does.--> |
| 331 | <!--<param name="NDLB-force-rport" value="true"/>--> |
| 332 | <!-- |
| 333 | Choose the realm challenge key. Default is auto_to if not set. |
| 334 | |
| 335 | auto_from - uses the from field as the value for the sip realm. |
| 336 | auto_to - uses the to field as the value for the sip realm. |
| 337 | <anyvalue> - you can input any value to use for the sip realm. |
| 338 | |
| 339 | If you want URL dialing to work you'll want to set this to auto_from. |
| 340 | |
| 341 | If you use any other value besides auto_to or auto_from you'll |
| 342 | loose the ability to do multiple domains. |
| 343 | |
| 344 | Note: comment out to restore the behavior before 2008-09-29 |
| 345 | --> |
| 346 | <param name="challenge-realm" value="auto_from"/> |
| 347 | <!--<param name="disable-rtp-auto-adjust" value="true"/>--> |
| 348 | <!-- on inbound calls make the uuid of the session equal to the sip call id of that call --> |
| 349 | <!--<param name="inbound-use-callid-as-uuid" value="true"/>--> |
| 350 | <!-- on outbound calls set the callid to match the uuid of the session --> |
| 351 | <!--<param name="outbound-use-uuid-as-callid" value="true"/>--> |
| 352 | <!-- set to false disable this feature --> |
| 353 | <!--<param name="rtp-autofix-timing" value="false"/>--> |
| 354 | |
| 355 | <!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore--> |
| 356 | <!--<param name="pass-callee-id" value="false"/>--> |
| 357 | |
| 358 | <!-- clear clears them all or supply the name to add or the name |
| 359 | prefixed with ~ to remove valid values: |
| 360 | |
| 361 | clear |
| 362 | CISCO_SKIP_MARK_BIT_2833 |
| 363 | SONUS_SEND_INVALID_TIMESTAMP_2833 |
| 364 | |
| 365 | --> |
| 366 | <!--<param name="auto-rtp-bugs" data="clear"/>--> |
| 367 | |
| 368 | <!-- the following can be used as workaround with bogus SRV/NAPTR records --> |
| 369 | <!--<param name="disable-srv" value="false" />--> |
| 370 | <!--<param name="disable-naptr" value="false" />--> |
| 371 | |
| 372 | <!-- The following can be used to fine-tune timers within sofia's transport layer |
| 373 | Those settings are for advanced users and can safely be left as-is --> |
| 374 | |
| 375 | <!-- Initial retransmission interval (in milliseconds). |
| 376 | Set the T1 retransmission interval used by the SIP transaction engine. |
| 377 | The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. --> |
| 378 | <!-- <param name="timer-T1" value="500" /> --> |
| 379 | |
| 380 | <!-- Transaction timeout (defaults to T1 * 64). |
| 381 | Set the T1x64 timeout value used by the SIP transaction engine. |
| 382 | The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine. |
| 383 | The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. --> |
| 384 | <!-- <param name="timer-T1X64" value="32000" /> --> |
| 385 | |
| 386 | |
| 387 | <!-- Maximum retransmission interval (in milliseconds). |
| 388 | Set the maximum retransmission interval used by the SIP transaction engine. |
| 389 | The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine. |
| 390 | Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially |
| 391 | until the timer B fires. --> |
| 392 | <!-- <param name="timer-T2" value="4000" /> --> |
| 393 | |
| 394 | <!-- |
| 395 | Transaction lifetime (in milliseconds). |
| 396 | Set the lifetime for completed transactions used by the SIP transaction engine. |
| 397 | A completed transaction is kept around for the duration of T4 in order to catch late responses. |
| 398 | The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. --> |
| 399 | <!-- <param name="timer-T4" value="4000" /> --> |
| 400 | |
| 401 | <!-- Turn on a jitterbuffer for every call --> |
| 402 | <!-- <param name="auto-jitterbuffer-msec" value="60"/> --> |
| 403 | |
| 404 | |
| 405 | <!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations |
| 406 | Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold. |
| 407 | It's probably not what you want so stick with the default unless you really need to change this. |
| 408 | --> |
| 409 | <!--<param name="renegotiate-codec-on-hold" value="true"/>--> |
| 410 | |
| 411 | </settings> |
| 412 | </profile> |