diff --git a/UPGRADE.txt b/UPGRADE.txt index 8097c4bae75c987d45bdd9c298b7270cf3ac0aec..20d05807df42ef07d12f890820e43943e4a0b874 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -41,6 +41,17 @@ From 1.6.2 to 1.6.3: From 1.6.1 to 1.6.2: +* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers + has been renamed to 'directmedia', to better reflect what it actually does. + In the case of SIP, there are still re-INVITEs issued for T.38 negotiation, + starting and stopping music-on-hold, and other reasons, and the 'canreinvite' + option never had any effect on these cases, it only affected the re-INVITEs + used for direct media path setup. For MGCP and Skinny, the option was poorly + named because those protocols don't even use INVITE messages at all. For + backwards compatibility, the old option is still supported in both normal + and Realtime configuration files, but all of the sample configuration files, + Realtime/LDAP schemas, and other documentation refer to it using the new name. + * The default console now will use colors according to the default background color, instead of forcing the background color to black. If you are using a light colored background for your console, you may wish to use the option diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c index d682c76bb10cbafc2dc15308df127e86d5de98a1..8adaf12a356b302c385b07f4de276be4e8ec012b 100644 --- a/channels/chan_mgcp.c +++ b/channels/chan_mgcp.c @@ -80,7 +80,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #define MGCPDUMPER #define DEFAULT_EXPIRY 120 #define MAX_EXPIRY 3600 -#define CANREINVITE 1 +#define DIRECTMEDIA 1 #ifndef INADDR_NONE #define INADDR_NONE (in_addr_t)(-1) @@ -177,7 +177,7 @@ static int cancallforward = 0; static int singlepath = 0; -static int canreinvite = CANREINVITE; +static int directmedia = DIRECTMEDIA; static char accountcode[AST_MAX_ACCOUNT_CODE] = ""; @@ -330,7 +330,7 @@ struct mgcp_endpoint { int threewaycalling; int singlepath; int cancallforward; - int canreinvite; + int directmedia; int callreturn; int dnd; /* How does this affect callwait? Do we just deny a mgcp_request if we're dnd? */ int hascallerid; @@ -3552,7 +3552,7 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v) int i=0, y=0; int gw_reload = 0; int ep_reload = 0; - canreinvite = CANREINVITE; + directmedia = DIRECTMEDIA; /* locate existing gateway */ gw = gateways; @@ -3662,8 +3662,8 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v) cancallforward = ast_true(v->value); } else if (!strcasecmp(v->name, "singlepath")) { singlepath = ast_true(v->value); - } else if (!strcasecmp(v->name, "canreinvite")) { - canreinvite = ast_true(v->value); + } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) { + directmedia = ast_true(v->value); } else if (!strcasecmp(v->name, "mailbox")) { ast_copy_string(mailbox, v->value, sizeof(mailbox)); } else if (!strcasecmp(v->name, "hasvoicemail")) { @@ -3748,7 +3748,7 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v) e->callreturn = callreturn; e->cancallforward = cancallforward; e->singlepath = singlepath; - e->canreinvite = canreinvite; + e->directmedia = directmedia; e->callwaiting = callwaiting; e->hascallwaiting = callwaiting; e->slowsequence = slowsequence; @@ -3851,7 +3851,7 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v) e->pickupgroup=cur_pickupgroup; e->callreturn = callreturn; e->cancallforward = cancallforward; - e->canreinvite = canreinvite; + e->directmedia = directmedia; e->singlepath = singlepath; e->callwaiting = callwaiting; e->hascallwaiting = callwaiting; @@ -3944,7 +3944,7 @@ static enum ast_rtp_glue_result mgcp_get_rtp_peer(struct ast_channel *chan, stru *instance = sub->rtp ? ao2_ref(sub->rtp, +1), sub->rtp : NULL; - if (sub->parent->canreinvite) + if (sub->parent->directmedia) return AST_RTP_GLUE_RESULT_REMOTE; else return AST_RTP_GLUE_RESULT_LOCAL; diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 5e777b4c3f4b30b25d8c42d56c4442f8ebe1e025..78b42c8219ff0fda602f4694dc5f5e30c3d54c04 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1457,8 +1457,8 @@ struct sip_auth { /* re-INVITE related settings */ #define SIP_REINVITE (7 << 20) /*!< DP: four settings, uses three bits */ #define SIP_REINVITE_NONE (0 << 20) /*!< DP: no reinvite allowed */ -#define SIP_CAN_REINVITE (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */ -#define SIP_CAN_REINVITE_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */ +#define SIP_DIRECT_MEDIA (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */ +#define SIP_DIRECT_MEDIA_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */ #define SIP_REINVITE_UPDATE (4 << 20) /*!< DP: use UPDATE (RFC3311) when reinviting this peer */ /* "insecure" settings - see insecure2str() */ @@ -15369,7 +15369,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct ast_cli(fd, " Force rport : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT))); ast_cli(fd, " ACL : %s\n", cli_yesno(peer->ha != NULL)); ast_cli(fd, " T38 pt UDPTL : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT))); - ast_cli(fd, " CanReinvite : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE))); + ast_cli(fd, " DirectMedia : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA))); ast_cli(fd, " PromiscRedir : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR))); ast_cli(fd, " User=Phone : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE))); ast_cli(fd, " Video Support: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT))); @@ -15473,7 +15473,8 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE))); astman_append(s, "SIP-Forcerport: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT)?"Y":"N")); astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N")); - astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)?"Y":"N")); + astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N")); + astman_append(s, "SIP-DirectMedia: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N")); astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N")); astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N")); astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N")); @@ -23488,11 +23489,11 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP); ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP); } - } else if (!strcasecmp(v->name, "canreinvite")) { + } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) { ast_set_flag(&mask[0], SIP_REINVITE); ast_clear_flag(&flags[0], SIP_REINVITE); if (ast_true(v->value)) { - ast_set_flag(&flags[0], SIP_CAN_REINVITE | SIP_CAN_REINVITE_NAT); + ast_set_flag(&flags[0], SIP_DIRECT_MEDIA | SIP_DIRECT_MEDIA_NAT); } else if (!ast_false(v->value)) { char buf[64]; char *word, *next = buf; @@ -23500,12 +23501,12 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask ast_copy_string(buf, v->value, sizeof(buf)); while ((word = strsep(&next, ","))) { if (!strcasecmp(word, "update")) { - ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_CAN_REINVITE); + ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_DIRECT_MEDIA); } else if (!strcasecmp(word, "nonat")) { - ast_set_flag(&flags[0], SIP_CAN_REINVITE); - ast_clear_flag(&flags[0], SIP_CAN_REINVITE_NAT); + ast_set_flag(&flags[0], SIP_DIRECT_MEDIA); + ast_clear_flag(&flags[0], SIP_DIRECT_MEDIA_NAT); } else { - ast_log(LOG_WARNING, "Unknown canreinvite mode '%s' on line %d\n", v->value, v->lineno); + ast_log(LOG_WARNING, "Unknown directmedia mode '%s' on line %d\n", v->value, v->lineno); } } } @@ -24594,7 +24595,7 @@ static int reload_config(enum channelreloadreason reason) ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest)); ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten)); ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */ - ast_set_flag(&global_flags[0], SIP_CAN_REINVITE); /*!< Allow re-invites */ + ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */ ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine)); /* Debugging settings, always default to off */ @@ -25291,7 +25292,7 @@ static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan) return NULL; sip_pvt_lock(p); - if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) + if (p->udptl && ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) udptl = p->udptl; sip_pvt_unlock(p); return udptl; @@ -25342,7 +25343,7 @@ static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struc ao2_ref(p->rtp, +1); *instance = p->rtp; - if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE | SIP_CAN_REINVITE_NAT)) { + if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA | SIP_DIRECT_MEDIA_NAT)) { res = AST_RTP_GLUE_RESULT_REMOTE; } else if (ast_test_flag(&global_jbconf, AST_JB_FORCED)) { res = AST_RTP_GLUE_RESULT_FORBID; @@ -25371,7 +25372,7 @@ static enum ast_rtp_glue_result sip_get_vrtp_peer(struct ast_channel *chan, stru ao2_ref(p->vrtp, +1); *instance = p->vrtp; - if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) { + if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) { res = AST_RTP_GLUE_RESULT_REMOTE; } @@ -25398,7 +25399,7 @@ static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, stru ao2_ref(p->trtp, +1); *instance = p->trtp; - if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) { + if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) { res = AST_RTP_GLUE_RESULT_REMOTE; } @@ -25430,7 +25431,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i /* if this peer cannot handle reinvites of the media stream to devices that are known to be behind a NAT, then stop the process now */ - if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) { + if (nat_active && !ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) { sip_pvt_unlock(p); return 0; } diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c index 6a82c74e56ecd808a2d6ad0dada04e96e93fbe15..edaf148d80ce62d60a48406b343e047a138fd706 100644 --- a/channels/chan_skinny.c +++ b/channels/chan_skinny.c @@ -1239,7 +1239,7 @@ struct skinny_subchannel { int immediate; \ int hookstate; \ int nat; \ - int canreinvite; \ + int directmedia; \ int prune; struct skinny_line { @@ -1265,7 +1265,7 @@ static struct skinny_line_options{ .hidecallerid = 0, .amaflags = 0, .instance = 0, - .canreinvite = 0, + .directmedia = 0, .nat = 0, .confcapability = AST_FORMAT_ULAW | AST_FORMAT_ALAW, .capability = 0, @@ -2689,7 +2689,7 @@ static enum ast_rtp_glue_result skinny_get_rtp_peer(struct ast_channel *c, struc l = sub->parent; - if (!l->canreinvite || l->nat){ + if (!l->directmedia || l->nat){ res = AST_RTP_GLUE_RESULT_LOCAL; if (skinnydebug) ast_verb(1, "skinny_get_rtp_peer() Using AST_RTP_GLUE_RESULT_LOCAL \n"); @@ -2749,7 +2749,7 @@ static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp_instance *r req->data.startmedia.conferenceId = htolel(sub->callid); req->data.startmedia.passThruPartyId = htolel(sub->callid); - if (!(l->canreinvite) || (l->nat)){ + if (!(l->directmedia) || (l->nat)){ ast_rtp_instance_get_local_address(rtp, &us); req->data.startmedia.remoteIp = htolel(d->ourip.s_addr); req->data.startmedia.remotePort = htolel(ntohs(us.sin_port)); @@ -6717,9 +6717,9 @@ static struct ast_channel *skinny_request(const char *type, int format, const st CLINE_OPTS->callwaiting = ast_true(v->value); continue; } - } else if (!strcasecmp(v->name, "canreinvite")) { + } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) { if (type & (TYPE_DEF_LINE | TYPE_LINE)) { - CLINE_OPTS->canreinvite = ast_true(v->value); + CLINE_OPTS->directmedia = ast_true(v->value); continue; } } else if (!strcasecmp(v->name, "nat")) { diff --git a/configs/mgcp.conf.sample b/configs/mgcp.conf.sample index 116b66cd036eb606117fb334901b603c2cb32a4e..fde0a4fc6daa90d192e3a80c9a50256812fc74ba 100644 --- a/configs/mgcp.conf.sample +++ b/configs/mgcp.conf.sample @@ -41,7 +41,7 @@ ;[dlinkgw] ;host = 192.168.0.64 ;context = default -;canreinvite = no +;directmedia = no ;line => aaln/2 ;line => aaln/1 @@ -96,7 +96,7 @@ ;callwaiting = no ;callreturn = yes ;cancallforward = yes -;canreinvite = no +;directmedia = no ;transfer = no ;dtmfmode = inband ;line => aaln/1 ; now lets save this config to line1 aka aaln/1 @@ -104,7 +104,7 @@ ;callwaiting = no ;callreturn = yes ;cancallforward = yes -;canreinvite = no +;directmedia = no ;transfer = no ;dtmfmode = inband ;line => aaln/2 ; now lets save this config to line2 aka aaln/2 diff --git a/configs/res_ldap.conf.sample b/configs/res_ldap.conf.sample index 0a442298da78a95b0b66a278c7381a0621b985a4..b02045f15c71e85352c4e00194c63de705954719 100644 --- a/configs/res_ldap.conf.sample +++ b/configs/res_ldap.conf.sample @@ -60,7 +60,7 @@ name = cn amaflags = AstAccountAMAFlags callgroup = AstAccountCallGroup callerid = AstAccountCallerID -canreinvite = AstAccountCanReinvite +directmedia = AstAccountDirectMedia context = AstAccountContext dtmfmode = AstAccountDTMFMode fromuser = AstAccountFromUser @@ -131,7 +131,7 @@ additionalFilter=(objectClass=*) amaflags = AstAccountAMAFlags callgroup = AstAccountCallGroup callerid = AstAccountCallerID -canreinvite = AstAccountCanReinvite +directmedia = AstAccountDirectMedia context = AstAccountContext dtmfmode = AstAccountDTMFMode fromuser = AstAccountFromUser diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index fef5ef8f424bd7f637ee36319b6740516f185350..ba9b0c619f8e174302391153f63f700aec80352d 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -662,17 +662,17 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; nat = comedia ; Use rport if the remote side says to use it and perform symmetric RTP. ;----------------------------------- MEDIA HANDLING -------------------------------- -; By default, Asterisk tries to re-invite the audio to an optimal path. If there's +; By default, Asterisk tries to re-invite media streams to an optimal path. If there's ; no reason for Asterisk to stay in the media path, the media will be redirected. -; This does not really work with in the case where Asterisk is outside and have -; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat +; This does not really work well in the case where Asterisk is outside and the +; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat. ; -;canreinvite=yes ; Asterisk by default tries to redirect the - ; RTP media stream (audio) to go directly from +;directmedia=yes ; Asterisk by default tries to redirect the + ; RTP media stream to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is behind a NAT). ; The default setting is YES. If you have all clients - ; behind a NAT, or for some other reason wants Asterisk to + ; behind a NAT, or for some other reason want Asterisk to ; stay in the audio path, you may want to turn this off. ; This setting also affect direct RTP @@ -684,18 +684,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; the call directly with media peer-2-peer without re-invites. ; Will not work for video and cases where the callee sends ; RTP payloads and fmtp headers in the 200 OK that does not match the - ; callers INVITE. This will also fail if canreinvite is enabled when + ; callers INVITE. This will also fail if directmedia is enabled when ; the device is actually behind NAT. -;canreinvite=nonat ; An additional option is to allow media path redirection +;directmedia=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being ; sent is known to not be behind a NAT (as the RTP core can ; determine it based on the apparent IP address the media ; arrives from). -;canreinvite=update ; Yet a third option... use UPDATE for media path redirection, +;directmedia=update ; Yet a third option... use UPDATE for media path redirection, ; instead of INVITE. This can be combined with 'nonat', as - ; 'canreinvite=update,nonat'. It implies 'yes'. + ; 'directmedia=update,nonat'. It implies 'yes'. ;ignoresdpversion=yes ; By default, Asterisk will honor the session version ; number in SDP packets and will only modify the SDP @@ -859,7 +859,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; remotesecret ; transport ; dtmfmode -; canreinvite +; directmedia ; nat ; callgroup ; pickupgroup @@ -969,12 +969,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls [natted-phone](!,basic-options) ; another template inheriting basic-options nat=yes - canreinvite=no + directmedia=no host=dynamic [public-phone](!,basic-options) ; another template inheriting basic-options nat=no - canreinvite=yes + directmedia=yes [my-codecs](!) ; a template for my preferred codecs disallow=all @@ -1009,7 +1009,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;host=192.168.0.23 ; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk -;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk +;directmedia=yes ; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk (deprecated) @@ -1039,7 +1039,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;callerid="Jane Smith" <5678> ;host=dynamic ; This device needs to register ;nat=yes ; X-Lite is behind a NAT router -;canreinvite=no ; Typically set to NO if behind NAT +;directmedia=no ; Typically set to NO if behind NAT ;disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw @@ -1112,7 +1112,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Send SIP and RTP to the IP address that packet is ; received from instead of trusting SIP headers ;host=dynamic ; This device registers with us -;canreinvite=no ; Asterisk by default tries to redirect the +;directmedia=no ; Asterisk by default tries to redirect the ; RTP media stream (audio) to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample index 701723923d3493f9d8bf3740d12e54f316cdb557..dfbddd4c5ca04d759dd7fb2733821eaab53945ef 100644 --- a/configs/skinny.conf.sample +++ b/configs/skinny.conf.sample @@ -157,7 +157,7 @@ keepalive=120 ;device=SEP00D0BA847E6B ;version=P002G204 ; Thanks critch ;context=did -;canreinvite=yes ; Allow media to go directly between two RTP endpoints. +;directmedia=yes ; Allow media to go directly between two RTP endpoints. ;line=120 ; Dial(Skinny/120@florian) ; Typical config for a 7910 diff --git a/contrib/scripts/realtime_pgsql.sql b/contrib/scripts/realtime_pgsql.sql index 5539bbeb8d4ae59585b6b3a7d920c1eb3943051a..d50dff55b7aabf14cb08e9e8f30756746f229f98 100644 --- a/contrib/scripts/realtime_pgsql.sql +++ b/contrib/scripts/realtime_pgsql.sql @@ -37,7 +37,7 @@ accountcode character varying(20), amaflags character varying(7), callgroup character varying(10), callerid character varying(80), -canreinvite character varying(3) DEFAULT 'yes', +directmedia character varying(3) DEFAULT 'yes', context character varying(80), defaultip character varying(15), dtmfmode character varying(7), diff --git a/doc/chan_sip-perf-testing.txt b/doc/chan_sip-perf-testing.txt index 85b22bddc76d590f09a358a137db87b0e3bcce00..56992ac7fc668bd682dbb1267fb6dc5ad5d0e789 100644 --- a/doc/chan_sip-perf-testing.txt +++ b/doc/chan_sip-perf-testing.txt @@ -58,7 +58,7 @@ type=friend context=test11 host=192.168.134.240 ;; the address of the host you will be running sipp on user=sipp -canreinvite=no +directmedia=no disallow=all allow=ulaw diff --git a/doc/res_config_sqlite.txt b/doc/res_config_sqlite.txt index 39d31521af3c7c03e0ca21df6b164fbb1dc6bff7..95322cf1058462fab8d358ea86286af3a1fb80ee 100644 --- a/doc/res_config_sqlite.txt +++ b/doc/res_config_sqlite.txt @@ -70,7 +70,7 @@ CREATE TABLE ast_sip ( callgroup VARCHAR(10) DEFAULT NULL, callerid VARCHAR(80) DEFAULT NULL, cancallforward CHAR(3) DEFAULT 'yes', - canreinvite CHAR(3) DEFAULT 'yes', + directmedia CHAR(3) DEFAULT 'yes', context VARCHAR(80) DEFAULT NULL, defaultip VARCHAR(15) DEFAULT NULL, dtmfmode VARCHAR(7) DEFAULT NULL, diff --git a/doc/tex/phoneprov.tex b/doc/tex/phoneprov.tex index cb236a89acf113c4f63f1c9dba36f19928585c85..04ca22fd0e0b76476c571f10041f700ad14e9856 100644 --- a/doc/tex/phoneprov.tex +++ b/doc/tex/phoneprov.tex @@ -146,7 +146,7 @@ threewaycalling = yes deletevoicemail = no autoprov = yes profile = polycom -canreinvite = no +directmedia = no nat = no fullname = User Two ; ${DISPLAY_NAME} secret = test ; ${SECRET}