canceller on the channel (if any), for the current call
only.</para>
<para>Possible values are:</para>
- <para> <literal>on</literal> Normal mode (the echo canceller is actually reinitalized)</para>
+ <para> <literal>on</literal> Normal mode (the echo canceller is actually reinitialized)</para>
<para> <literal>off</literal> Disabled</para>
<para> <literal>fax</literal> FAX/data mode (NLP disabled if possible, otherwise
completely disabled)</para>
*
* \details
* original dialstring:
- * DAHDI/[i<span>-](g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension[/options]]
+ * DAHDI/[i<span>-](g|G|r|R)<group#(0-63)>[c|r<cadence#>|d][/extension[/options]]
*
* The modified dialstring will have prefixed the channel-group section
* with the ISDN channel restriction.
*
* buf:
- * DAHDI/i<span>-(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension[/options]]
+ * DAHDI/i<span>-(g|G|r|R)<group#(0-63)>[c|r<cadence#>|d][/extension[/options]]
*
* The routine will check to see if the ISDN channel restriction is already
* in the original dialstring.
*
* \param pvt Private channel structure.
* \param state Initial state of new channel.
- * \param law Combanding law to use.
+ * \param law Companding law to use.
* \param exten Dialplan extension for incoming call.
* \param requestor Channel requesting this new channel.
*
case SIG_FEATDMF:
return "Feature Group D (MF)";
case SIG_FEATDMF_TA:
- return "Feature Groud D (MF) Tandem Access";
+ return "Feature Group D (MF) Tandem Access";
case SIG_FEATB:
return "Feature Group B (MF)";
case SIG_E911:
ast_debug(1, "Normal call hung up with both three way call and a call waiting call in place?\n");
if (p->subs[SUB_CALLWAIT].inthreeway) {
/* We had flipped over to answer a callwait and now it's gone */
- ast_debug(1, "We were flipped over to the callwait, moving back and unowning.\n");
+ ast_debug(1, "We were flipped over to the callwait, moving back and not owning.\n");
/* Move to the call-wait, but un-own us until they flip back. */
swap_subs(p, SUB_CALLWAIT, SUB_REAL);
unalloc_sub(p, SUB_CALLWAIT);
int rr_starting_point;
/*! ISDN span where channels can be picked (Zero if not specified) */
int span;
- /*! Analog channel distinctive ring cadance index. */
- int cadance;
+ /*! Analog channel distinctive ring cadence index. */
+ int cadence;
/*! Dialing option. c/r/d if present and valid. */
char opt;
/*! TRUE if to search the channel list backwards. */
/*
* data is ---v
* Dial(DAHDI/pseudo[/extension[/options]])
- * Dial(DAHDI/<channel#>[c|r<cadance#>|d][/extension[/options]])
- * Dial(DAHDI/<subdir>!<channel#>[c|r<cadance#>|d][/extension[/options]])
+ * Dial(DAHDI/<channel#>[c|r<cadence#>|d][/extension[/options]])
+ * Dial(DAHDI/<subdir>!<channel#>[c|r<cadence#>|d][/extension[/options]])
* Dial(DAHDI/i<span>[/extension[/options]])
- * Dial(DAHDI/[i<span>-](g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension[/options]])
+ * Dial(DAHDI/[i<span>-](g|G|r|R)<group#(0-63)>[c|r<cadence#>|d][/extension[/options]])
*
* i - ISDN span channel restriction.
* Used by CC to ensure that the CC recall goes out the same span.
* R - channel group allocation round robin search backward
*
* c - Wait for DTMF digit to confirm answer
- * r<cadance#> - Set distintive ring cadance number
+ * r<cadence#> - Set distinctive ring cadence number
* d - Force bearer capability for ISDN/SS7 call to digital.
*/
if (toupper(args.group[0]) == 'G' || toupper(args.group[0])=='R') {
/* Retrieve the group number */
s = args.group + 1;
- res = sscanf(s, "%30d%1c%30d", &x, ¶m->opt, ¶m->cadance);
+ res = sscanf(s, "%30d%1c%30d", &x, ¶m->opt, ¶m->cadence);
if (res < 1) {
ast_log(LOG_WARNING, "Unable to determine group for data %s\n", data);
return NULL;
x = CHAN_PSEUDO;
param->channelmatch = x;
} else {
- res = sscanf(s, "%30d%1c%30d", &x, ¶m->opt, ¶m->cadance);
+ res = sscanf(s, "%30d%1c%30d", &x, ¶m->opt, ¶m->cadence);
if (res < 1) {
ast_log(LOG_WARNING, "Unable to determine channel for data %s\n", data);
return NULL;
break;
case 'r':
/* Distinctive ring */
- p->distinctivering = start.cadance;
+ p->distinctivering = start.cadence;
break;
case 'd':
#if defined(HAVE_PRI) || defined(HAVE_SS7)
return -1;
}
if (finish < start) {
- ast_log(LOG_WARNING, "Sillyness: %d < %d\n", start, finish);
+ ast_log(LOG_WARNING, "Silliness: %d < %d\n", start, finish);
x = finish;
finish = start;
start = x;
/*!
* * \brief Another container of iax2_pvt structures
*
- * Active IAX2 pvt stucts used during transfering a call are stored here.
+ * Active IAX2 pvt structs used during transfering a call are stored here.
*/
static struct ao2_container *iax_transfercallno_pvts;
* Container locked here since peercnt may be unlinked from
* list. If left unlocked, peercnt_add could try and grab this
* entry from the table and modify it at the "same time" this
- * thread attemps to unlink it.
+ * thread attempts to unlink it.
*/
ao2_lock(peercnts);
peercnt->cur--;
ast_update_realtime("iaxpeers", "name", peername,
"ipaddr", ast_sockaddr_isnull(sockaddr) ? "" : ast_sockaddr_stringify_addr(sockaddr),
"port", ast_sockaddr_isnull(sockaddr) ? "" : port,
- "regseconds", regseconds, syslabel, sysname, SENTINEL); /* note syslable can be NULL */
+ "regseconds", regseconds, syslabel, sysname, SENTINEL); /* note syslabel can be NULL */
}
struct create_addr_info {
remove_by_peercallno(pvt);
}
pvt->peercallno = peercallno;
- /*this is where the transfering call swiches hash tables */
+ /*this is where the transfering call switches hash tables */
store_by_peercallno(pvt);
pvt->transferring = TRANSFER_NONE;
pvt->svoiceformat = -1;
return send_command(iaxs[callno], AST_FRAME_IAX, IAX_COMMAND_REGREQ, 0, ied.buf, ied.pos, -1);
} else
return -1;
- ast_log(LOG_WARNING, "Registry acknowledge on unknown registery '%s'\n", peer);
+ ast_log(LOG_WARNING, "Registry acknowledge on unknown registry '%s'\n", peer);
} else
ast_log(LOG_NOTICE, "Can't reregister without a reg\n");
return -1;
res = send_trunk(tpeer, &now);
trunk_timed++;
if (iaxtrunkdebug) {
- ast_verbose(" - Trunk peer (%s) has %d call chunk%s in transit, %u bytes backloged and has hit a high water mark of %u bytes\n",
+ ast_verbose(" - Trunk peer (%s) has %d call chunk%s in transit, %u bytes backlogged and has hit a high water mark of %u bytes\n",
ast_sockaddr_stringify(&tpeer->addr),
res,
(res != 1) ? "s" : "",
char name[80];
struct mgcp_subchannel *sub; /*!< Pointer to our current connection, channel and stuff */
char accountcode[AST_MAX_ACCOUNT_CODE];
- char exten[AST_MAX_EXTENSION]; /*!< Extention where to start */
+ char exten[AST_MAX_EXTENSION]; /*!< Extension where to start */
char context[AST_MAX_EXTENSION];
char language[MAX_LANGUAGE];
char cid_num[AST_MAX_EXTENSION]; /*!< Caller*ID number */
timeout = firstdigittimeout;
} else if (!strcmp(p->dtmf_buf, pickupexten)) {
/* Scan all channels and see if any there
- * ringing channqels with that have call groups
+ * ringing channels with that have call groups
* that equal this channels pickup group
*/
if (ast_pickup_call(chan)) {
sub->cxmode = MGCP_CX_SENDRECV;
if (p) {
- /* When the endpoint have a Off hook transition we allways
+ /* When the endpoint have a Off hook transition we always
starts without any previous dtmfs */
memset(p->dtmf_buf, 0, sizeof(p->dtmf_buf));
}
}
*/
if (p->transfer && (sub->owner && sub->next->owner) && ((!sub->outgoing) || (!sub->next->outgoing))) {
- /* We're allowed to transfer, we have two avtive calls and */
+ /* We're allowed to transfer, we have two active calls and */
/* we made at least one of the calls. Let's try and transfer */
ast_mutex_lock(&p->sub->next->lock);
res = attempt_transfer(p, sub);
<synopsis>Maximum number of ICE candidates to offer</synopsis>
</configOption>
<configOption name="maxpayloads">
- <synopsis>Maximum number of pyaloads to offer</synopsis>
+ <synopsis>Maximum number of payloads to offer</synopsis>
</configOption>
</configObject>
</configFile>
rtp = session->vrtp;
}
} else {
- /* Google-V1 has no concept of assocating things like the above does, so since we only support audio over it assume they want audio */
+ /* Google-V1 has no concept of associating things like the above does, so since we only support audio over it assume they want audio */
rtp = session->rtp;
}
/*! \file
*
* \author Joshua Colp <jcolp@digium.com>
- * \author Andreas 'MacBrody' Bro
admann <andreas.brodmann@gmail.com>
+ * \author Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
*
* \brief RTP (Multicast and Unicast) Media Channel
*
static char default_language[MAX_LANGUAGE]; /*!< Default language setting for new channels */
static char default_callerid[AST_MAX_EXTENSION]; /*!< Default caller ID for sip messages */
static char default_mwi_from[80]; /*!< Default caller ID for MWI updates */
-static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outound messages */
+static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outbound messages */
static int default_fromdomainport; /*!< Default domain port on outbound messages */
static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
static char default_vmexten[AST_MAX_EXTENSION]; /*!< Default From Username on MWI updates */
Without a dialog we can't retransmit and handle ACKs and all that, but at least
send an error message.
- Sorry, we apologize for the inconvienience
+ Sorry, we apologize for the inconvenience
*/
transmit_response_using_temp(callid, addr, 1, intended_method, req, "500 Server internal error");
ast_debug(4, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n");
}
/*!
- \brief Choose realm based on From header and then To header or use globaly configured realm.
+ \brief Choose realm based on From header and then To header or use globally configured realm.
Realm from From/To header should be listed among served domains in config file: domain=...
*/
static void get_realm(struct sip_pvt *p, const struct sip_request *req)
if (doing_directmedia) {
ast_format_cap_get_compatible(p->jointcaps, p->redircaps, tmpcap);
- ast_debug(1, "** Our native-bridge filtered capablity: %s\n", ast_format_cap_get_names(tmpcap, &codec_buf));
+ ast_debug(1, "** Our native-bridge filtered capability: %s\n", ast_format_cap_get_names(tmpcap, &codec_buf));
} else {
ast_format_cap_append_from_cap(tmpcap, p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
}
case 200: /* OK: The new call is up, hangup this call */
/* Hangup the call that we are replacing */
break;
- case 301: /* Moved permenantly */
+ case 301: /* Moved permanently */
case 302: /* Moved temporarily */
/* Do we get the header in the packet in this case? */
success = FALSE;
info->summary = "SIP TCP message fragmentation test";
info->description =
"Tests reception of different TCP messages that have been fragmented or"
- "run together. This test mimicks the code that TCP reception uses.";
+ "run together. This test mimics the code that TCP reception uses.";
return AST_TEST_NOT_RUN;
case TEST_EXECUTE:
break;
static int matchdigittimeout = 3000;
/*!
- * To apease the stupid compiler option on ast_sched_del()
+ * To appease the stupid compiler option on ast_sched_del()
* since we don't care about the return value.
*/
static int not_used;
return;
//what do we want hear CLEAR_DISPLAY_MESSAGE or CLEAR_PROMPT_STATUS???
- //if we are clearing the display, it appears there is no instance and refernece info (size 0)
+ //if we are clearing the display, it appears there is no instance and reference info (size 0)
//req->data.clearpromptstatus.lineInstance = instance;
//req->data.clearpromptstatus.callReference = reference;
else
req->data.forwardstat.activeforward = htolel(0);
- SKINNY_DEBUG(DEBUG_PACKET, 3, "Transmitting FORWARD_STAT_MESSAGE to %s, inst %d, all %s, busy %s, noans %s, acitve %d\n",
+ SKINNY_DEBUG(DEBUG_PACKET, 3, "Transmitting FORWARD_STAT_MESSAGE to %s, inst %d, all %s, busy %s, noans %s, active %d\n",
d->name, l->instance, l->call_forward_all, l->call_forward_busy, l->call_forward_noanswer, anyon ? 7 : 0);
transmit_response(d, req);
}
transmit_lamp_indication(d, STIMULUS_VOICEMAIL, l->instance, SKINNY_LAMP_OFF);
}
- /* find out wether the device lamp should be on or off */
+ /* find out whether the device lamp should be on or off */
AST_LIST_TRAVERSE(&d->lines, l2, list) {
if (l2->newmsgs) {
dev_msgs++;
unsigned short last_seq_ack; /*!< sequence number of the last ACK received */
unsigned long tick_next_ping; /*!< time for the next ping */
int last_buf_available; /*!< number of a free slot */
- int nb_retransmit; /*!< number of retransmition */
+ int nb_retransmit; /*!< number of retransmission */
int state; /*!< state of the phone (see phone_state) */
int size_buff_entry; /*!< size of the buffer used to enter datas */
char buff_entry[16]; /*!< Buffer for temporary datas */
/* ISO-8859-4 - Baltic) */
static const unsigned char packet_send_charset_iso_8859_4[] =
{ 0x17, 0x08, 0x21, 0x1b, 0x2d, 0x44, 0x1b, 0x00 };
-/* ISO 8859-5 - cyrilic */
+/* ISO 8859-5 - cyrillic */
static const unsigned char packet_send_charset_iso_8859_5[] =
{ 0x17, 0x08, 0x21, 0x1b, 0x2d, 0x4c, 0x1b, 0x00 };
-/* Japaneese (ISO-2022-JP ?) */
+/* Japanese (ISO-2022-JP ?) */
static const unsigned char packet_send_charset_iso_2022_jp[] =
{ 0x17, 0x08, 0x21, 0x1b, 0x29, 0x49, 0x1b, 0x7e };
sub = get_sub(s->device, SUB_THREEWAY);
if (sub) {
- /* If sub for threway call created than we use transfer behaviuor */
+ /* If sub for threway call created than we use transfer behavior */
struct unistim_subchannel *sub_trans = NULL;
struct unistim_device *d = s->device;
enable/disable Picture-in-Picture, freeze the incoming video,
dial numbers, pick up or hang up a call, ...)
-Configuration options control the appeareance of the gui:
+Configuration options control the appearance of the gui:
keypad = /tmp/kpad2.jpg ; the skin
keypad_font = /tmp/font.png ; the font to use for output
-For future implementation, intresting features can be the following:
+For future implementation, interesting features can be the following:
- save of the whole SDL window as a picture
- audio output device switching
* below the source windows
*/
-/* costants defined to describe status of devices */
+/* constants defined to describe status of devices */
#define IS_PRIMARY 1
#define IS_SECONDARY 2
#define IS_ON 4
button.x < x0+gui->keypad->w/2+BORDER+pip_loc_x+env->loc_dpy.w/3 &&
button.y >= BORDER+pip_loc_y &&
button.y < BORDER+pip_loc_y+env->loc_dpy.h/3) {
- /* set the y cordinate to his previous value */
+ /* set the y coordinate to his previous value */
button.y += (env->out.device_num ? SRC_WIN_H+2*BORDER+SRC_MSG_BD_H : 0);
/* starts dragging the picture inside the picture */
set_drag(&gui->drag, button.x, button.y, DRAG_PIP);
}
else if (index == KEY_LOC_DPY) {
- /* set the y cordinate to his previous value */
+ /* set the y coordinate to his previous value */
button.y += (env->out.device_num ? SRC_WIN_H+2*BORDER+SRC_MSG_BD_H : 0);
/* click in the local display, but not on the PiP */
set_drag(&gui->drag, button.x, button.y, DRAG_LOCAL);
static void keypad_setup(struct gui_info *gui, const char *kp_file);
-/* TODO: consistency checks, check for bpp, widht and height */
+/* TODO: consistency checks, check for bpp, width and height */
/* Init the mask image used to grab the action. */
static struct gui_info *gui_init(const char *keypad_file, const char *font)
{
/*
* Codecs are absolutely necessary or we cannot do anything.
* SDL is optional (used for rendering only), so that we can still
- * stream video withouth displaying it.
+ * stream video without displaying it.
*/
#if !defined(HAVE_VIDEO_CONSOLE) || !defined(HAVE_FFMPEG)
/* stubs if required pieces are missing */
/*
* this structure will be an entry in the table containing
- * every device specified in the file oss.conf, it contains various infomation
+ * every device specified in the file oss.conf, it contains various information
* about the device
*/
struct video_device {
struct fbuf_t *dev_buf; /* buffer for incoming data */
struct timeval last_frame; /* when we read the last frame ? */
int status_index; /* what is the status of the device (source) */
- /* status index is set using the IS_ON, IS_PRIMARY and IS_SECONDARY costants */
+ /* status index is set using the IS_ON, IS_PRIMARY and IS_SECONDARY constants */
/* status_index is the index of the status message in the src_msgs array in console_gui.c */
};
* is returned as an argument.
*
* \param env = video environment descriptor
- * \param tail = tail ponter (pratically a return value)
+ * \param tail = tail ponter (practically a return value)
*/
static struct ast_frame *get_video_frames(struct video_desc *env, struct ast_frame **tail)
{
updating the private device buffer in the device table */
for (i = 0; i < env->out.device_num; i++) {
p_read = grabber_read(&env->out.devices[i], env->out.fps);
- /* it is used only if different from NULL, we mantain last good buffer otherwise */
+ /* it is used only if different from NULL, we maintain last good buffer otherwise */
if (p_read)
env->out.devices[i].dev_buf = p_read;
}
#if defined(HAVE_PRI)
/*
* PRI nobch channels (hold and call waiting) are equivalent to
- * pseudo channels and cannot be nativly bridged.
+ * pseudo channels and cannot be natively bridged.
*/
|| (dahdi_sig_pri_lib_handles(p0->sig)
&& ((struct sig_pri_chan *) p0->sig_pvt)->no_b_channel)
*
* <b>Sample Usage:</b>
* \code
- * struct sample_struct_componets {
+ * struct sample_struct_components {
* ASTOBJ_COMPONENTS_NOLOCK(struct sample_struct);
* };
* \endcode
/*!
* \internal
- * \brief Add firwmare related IEs to an IAX2 IE buffer.
+ * \brief Add firmware related IEs to an IAX2 IE buffer.
*
* \param ie_data The IE buffer being appended to.
* \param device_name The name of the requested firmware.
{ IAX_IE_TRANSFERID, "TRANSFER ID", dump_int },
{ IAX_IE_RDNIS, "REFERRING DNIS", dump_string },
{ IAX_IE_PROVISIONING, "PROVISIONING", dump_prov },
- { IAX_IE_AESPROVISIONING, "AES PROVISIONG" },
+ { IAX_IE_AESPROVISIONING, "AES PROVISIONING" },
{ IAX_IE_DATETIME, "DATE TIME", dump_datetime },
{ IAX_IE_DEVICETYPE, "DEVICE TYPE", dump_string },
{ IAX_IE_SERVICEIDENT, "SERVICE IDENT", dump_string },
{ IAX_IE_FIRMWAREVER, "FIRMWARE VER", dump_short },
{ IAX_IE_FWBLOCKDESC, "FW BLOCK DESC", dump_int },
{ IAX_IE_FWBLOCKDATA, "FW BLOCK DATA" },
- { IAX_IE_PROVVER, "PROVISIONG VER", dump_int },
+ { IAX_IE_PROVVER, "PROVISIONING VER", dump_int },
{ IAX_IE_CALLINGPRES, "CALLING PRESNTN", dump_byte },
{ IAX_IE_CALLINGTON, "CALLING TYPEOFNUM", dump_byte },
{ IAX_IE_CALLINGTNS, "CALLING TRANSITNET", dump_short },
cmd = "QUELCH ";
break;
case IAX_COMMAND_UNQUELCH:
- cmd = "UNQULCH";
+ cmd = "UNQUELCH";
break;
case IAX_COMMAND_POKE:
cmd = "POKE ";
ast_debug(1, "Normal call hung up with both three way call and a call waiting call in place?\n");
if (p->subs[ANALOG_SUB_CALLWAIT].inthreeway) {
/* We had flipped over to answer a callwait and now it's gone */
- ast_debug(1, "We were flipped over to the callwait, moving back and unowning.\n");
+ ast_debug(1, "We were flipped over to the callwait, moving back and not owning.\n");
/* Move to the call-wait, but un-own us until they flip back. */
analog_swap_subs(p, ANALOG_SUB_CALLWAIT, ANALOG_SUB_REAL);
analog_unalloc_sub(p, ANALOG_SUB_CALLWAIT);
int x;
if (principle < 0 || pri->numchans <= principle) {
- /* Out of rannge */
+ /* Out of range */
return -1;
}
if (!call) {
}
ast_frfree(f);
}
- /* Hangup the channel since nothing happend */
+ /* Hangup the channel since nothing happened */
ast_hangup(chan);
return NULL;
}
pri_find_dchan(pri);
}
- /* Note presense of D-channel */
+ /* Note presence of D-channel */
time(&pri->lastreset);
/* Restart in 5 seconds */
}
/*
* If hangup was delayed for this AOC-E msg, waiting_for_aoc
- * will be set. A hangup is already occuring via a timeout during
+ * will be set. A hangup is already occurring via a timeout during
* this delay. Instead of waiting for that timeout to occur, go ahead
* and initiate the hangup since the delay is no longer necessary.
*/
unsigned int hidecallerid:1;
unsigned int hidecalleridname:1; /*!< Hide just the name not the number for legacy PBX use */
unsigned int immediate:1; /*!< Answer before getting digits? */
- unsigned int priexclusive:1; /*!< Whether or not to override and use exculsive mode for channel selection */
+ unsigned int priexclusive:1; /*!< Whether or not to override and use exclusive mode for channel selection */
unsigned int priindication_oob:1;
unsigned int use_callerid:1; /*!< Whether or not to use caller id on this channel */
unsigned int use_callingpres:1; /*!< Whether to use the callingpres the calling switch sends */
unsigned int hidecallerid:1;
unsigned int hidecalleridname:1; /*!< Hide just the name not the number for legacy PBX use */
unsigned int immediate:1; /*!< Answer before getting digits? */
- unsigned int priexclusive:1; /*!< Whether or not to override and use exculsive mode for channel selection */
+ unsigned int priexclusive:1; /*!< Whether or not to override and use exclusive mode for channel selection */
unsigned int priindication_oob:1;
unsigned int use_callerid:1; /*!< Whether or not to use caller id on this channel */
unsigned int use_callingpres:1; /*!< Whether to use the callingpres the calling switch sends */
break;
}
p = linkset->pvts[chanpos];
- ast_debug(1, "Unequiped Circuit Id Code on CIC %d\n", e->ucic.cic);
+ ast_debug(1, "Unequipped Circuit Id Code on CIC %d\n", e->ucic.cic);
sig_ss7_lock_private(p);
sig_ss7_lock_owner(linkset, chanpos);
if (p->owner) {
SIG_SS7_ALAW
};
-enum sig_ss7_redirect_idication {
+enum sig_ss7_redirect_indication {
SS7_INDICATION_NO_REDIRECTION = 0,
SS7_INDICATION_REROUTED_PRES_ALLOWED,
SS7_INDICATION_REROUTED_INFO_RESTRICTED,
* \param flags An array of ast_flags that will be set by this function
*
* \note The nat-related values in both mask and flags are assumed to empty. This function
- * will treat the first "yes" or "no" value in a list of values as overiding all other values
+ * will treat the first "yes" or "no" value in a list of values as overriding all other values
* and will stop parsing. Auto values will override their non-auto counterparts.
*/
void sip_parse_nat_option(const char *value, struct ast_flags *mask, struct ast_flags *flags);
int parse_name_andor_addr(char *uri, const char *scheme, char **name,
char **user, char **pass, char **domain,
struct uriparams *params, char **headers,
- char **remander);
+ char **residue);
/*! \brief Parse all contact header contacts
* \retval 0 success
};
/*! \brief When sending a SIP message, we can send with a few options, depending on
- * type of SIP request. UNRELIABLE is moslty used for responses to repeated requests,
+ * type of SIP request. UNRELIABLE is mostly used for responses to repeated requests,
* where the original response would be sent RELIABLE in an INVITE transaction
*/
enum xmittype {
int t38_maxdatagram; /*!< T.38 FaxMaxDatagram override */
int request_queue_sched_id; /*!< Scheduler ID of any scheduled action to process queued requests */
int provisional_keepalive_sched_id; /*!< Scheduler ID for provisional responses that need to be sent out to avoid cancellation */
- const char *last_provisional; /*!< The last successfully transmitted provisonal response message */
+ const char *last_provisional; /*!< The last successfully transmitted provisional response message */
int authtries; /*!< Times we've tried to authenticate */
struct sip_proxy *outboundproxy; /*!< Outbound proxy for this dialog. Use ref_proxy to set this instead of setting it directly*/
struct t38properties t38; /*!< T38 settings */
/* clear any empty characters in the beginning */
input = ast_skip_blanks(input);
- /* make sure the output buffer is initilized */
+ /* make sure the output buffer is initialized */
*orig_output = '\0';
/* make room for '\0' at the end of the output buffer */
return first;
}
-/*! \brief extract the bitstreem from the RTP payload.
+/*! \brief extract the bitstream from the RTP payload.
* This is format dependent.
* For h263+, the format is defined in RFC 2429
* and basically has a fixed 2-byte header as follows:
projects / thirdparty / asterisk.git / commitdiff
