Dial()

Synopsis

Description

Parameters

• Jumping in Asterisk v1.2.14: In [general] you can set priorityjumping=yes/no. The default as of 1.2.14 is "yes". When set to "yes", the dialplan will jump to priority +101 on busy, congested, and channel unavailable. The wiki "used" to imply that the default was "no" if priorityjumping was not set. This does not appear to be the case in 1.2.14.
• New in Asterisk 1.2.0: If you don't want to modify options on each app that used to have jumping behavior, you can set "priorityjumping=yes" in the [general] section of extensions.conf which will enforce the old behavior globally. As far as the Dial() application is concerned you can control the behavior with the 'j' option (see below).
• New in Asterisk v1.2.0: The Caller*ID of the outbound leg is now the extension that was called, rather than the Caller*ID of the inbound leg of the call. The "o" flag for Dial can be used to restore the original behavior if desired. Note that if you are looking for the originating callerid from the manager event, there is a new manager event "Dial" which provides the source and destination channels and callerid.
• type specifies the channel type. It should be one of the registered channel types, such as "Zap", "SIP", "IAX2", and so on.
• identifier specifies the "phone number" to dial on that channel. The format of the "phone number" depends on the channel, and may contain additonal parameters (e.g. a distinctive ring parameter) specific to the channel module in question; the Dial command simply passes identifier to the channel module to process in whatever way is appropriate. See the documentation for the individual channel modules to learn about the correct format for specifying the identifier for the Dial command, and the options available to you when doing so. If you need a .5 second pause while dialing a number you can insert a w in the appropriate place.
• If you wish to specify more than one channel for the Dial command to try — remembering that it will dial out on all of them simultaneously — separate them with the & symbol (there must not be a space before and after the ampersand). The channels can be different types; see Examples, below. Note that call progress information is not consistantly handled with multiple channels.. consider using the 'r' option.
• The timeout parameter is optional. If not specifed, the Dial command will wait indefinitely, exiting only when the originating channel hangs up, or all the dialed channels return a busy or error condition. Otherwise it specifies a maximum time, in seconds, that the Dial command is to wait for a channel to answer.
• The options parameter, which is optional, is a string containging zero or more of the following flags and parameters:

• t: Allow the called user to transfer the call by hitting the blind xfer keys (features.conf)
  • If you have set the variable GOTO_ON_TRANSFER then the transferrer will be sent to the context|exten|pri (you can use ^ to represent | to avoid escapes), example: SetVar(GOTO_ON_TRANSFER=woohoo^s^1); works with both t and T
  • WARNING: GOTO_ON_TRANSFER does not exist in any version of ASTERISK and will not! the variable is called GOTO_ON_BLINDXFR see http://svn.digium.com/view/asterisk?rev=5495&view=rev and http://bugs.digium.com/view.php?id=4056 for details. THX to the person who shared the information above!
• T: Allow the calling user to transfer the call by hitting the blind xfer keys (features.conf)
• r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Without this option, Asterisk will generate ring tones automatically where it is appropriate to do so; however, "r" will force Asterisk to generate ring tones, even if it is not appropriate. For example, if you used this option to force ringing but the line was busy the user would hear "RING RIBEEP BEEP BEEP" (thank you tzanger), which is potentially confusing and/or unprofessional. However, the option is necessary in a couple of places. For example, when you're dialing multiple channels, call progress information is not consistantly passed back.
• R: Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. This is available only if you are using kapejod's Bristuff.
• m: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold.
• n: (Asterisk 1.1 and later) July 2005 bug 752 was included in CVS (Asterisk 1.1) and enhances the privacy manager considerably. As part of this patch, the 'n' flag to Dial got changed to be used as part of the privacy features, instead of being the 'dont jump to +101' flag. That flag is now 'j'.
• o: Restore the Asterisk v1.0 CallerId behaviour (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number)
• j: Asterisk 1.2 and later: Jump to priority n+101 if all of the requested channels were busy (just like behaviour in Asterisk 1.0.x)
• M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers). IMPORTANT - The CDR 'billsecs' field is set to zero if the callee answers the call, but hangs up whilst the macro is still running (if the callee answers and the macro finishes, 'billsecs' contains the correct value).
• h: Allow the callee to hang up by dialing *
• H: Allow the caller to hang up by dialing *
• i: Asterisk will ignore any forwarding requests it may receive on this dial attempt. (new in 1.4) Useful if you are ringing a group of people and one person has set their phone to forwarded direct to voicemail on their cell or something which normally prevents any of the other phones from ringing.
• C: Reset the CDR (Call Detail Record) for this call. This is like using the NoCDR command
• p: This option enables screening mode. This is basically Privacy mode without memory of how to handle the caller. It looks for the file sounds/priv-callerintros/${IF($[ "${CALLERID(num)}" != "" ]?${CALLERID(num)}:NOCALLERID_${EXTEN}${CUT(CHANNEL,/,1)}=${CUT(CHANNEL,/,2)})}.gsm and if it is not found, prompts the caller to say his name. It then rings the called party and plays sounds/priv-callpending, sounds/priv-callerintros/<see-above>, and sounds/screen-callee-options. If the called party enters 1, the call is accepted, 2, the DIAL command exits with ${DIALSTATUS} set to NOANSWER, 3, set to TORTURE and 4, set to DONTCALL. If no valid entry is made, the DIAL command exits with ${DIALSTATUS} set to ANSWER. The check for pre-existence of the name recording may not be what you want. For example, everyone from the same number is not necessarily the same person, especially if the number is OUTOFAREA, but if the file is there, no new name will be recorded. Since the files are never removed, you may wish to remove them with a System(rm /var/lib/asterisk/sounds/priv-callerintros/${IF($[ "${CALLERID(num)}" != "" ]?${CALLERID(num)}:NOCALLERID_${EXTEN}${CUT(CHANNEL,/,1)}=${CUT(CHANNEL,/,2)})}.*) right before the Dial command and clean up old ones with a cron job.
• P(x): Use the PrivacyManager, using x as the database (x is optional)
• g: When the called party hangs up, exit to execute more commands in the current context.
• G(context^exten^pri): If the call is answered, transfer both parties to the specified context and extension. The calling party is transferred to priority x, and the called party to priority x+1. This allows the dialplan to distinguish between the calling and called legs of the call (new in v1.2).
• A(x): Play an announcement (x.gsm) to the called party.
• S(n): Hangup the call n seconds AFTER called party picks up.
• d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also RetryDial
• D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.)
• L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c)
  • LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the caller.
  • LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee.
  • LIMIT_TIMEOUT_FILE - File to play when time is up.
  • LIMIT_CONNECT_FILE - File to play when call begins.
  • LIMIT_WARNING_FILE - File to play as warning if 'y' is defined. If LIMIT_WARNING_FILE is not defined, then the default behaviour is to announce ("You have [XX minutes] YY seconds").
• f: forces callerid to be set as the extension of the line making/redirecting the outgoing call. For example, some PSTNs don't allow callerids from other extensions than the ones that are assigned to you.
• w: Allow the called user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon)
• W: Allow the calling user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon)
• k: Allow the called party to enable parking of the call by sending the DTMF sequence defined for call parking in features.conf (Asterisk v1.4.x)
• K: Allow the calling party to enable parking of the call by sending the DTMF sequence defined for call parking in features.conf (Asterisk v1.4.x)

• The optional URL parameter will also be sent to the called party upon successful connection, if the channel technology supports the sending of URLs in this way.

Example 1: Simple use case

Dial macros

• Do not put spaces between the arguments to the Dial command, it will not work.
• When options t, T", "h", "H", "w", "W" or "L" (with multiple arguments) are applied, Asterisk will remain in the media path, even if canreinvite=yes'' (a SIP channel option) has been specified.

Return codes

• All channels dialed were busy
• There is exists something at n+101 in the current context
• You are running asterisk 1.0.x, priorityjumping=yes is set in extensions.conf, or the j option is specificed in the dial command

• DIALSTATUS - For more details see Asterisk Variable DIALSTATUS

• CHANUNAVAIL : Channel unavailable (for example in sip.conf, when using qualify=, the SIP chan is unavailable)
• BUSY : Returned busy
• NOANSWER : No Answer (i.e SIP 480 or 604 response)
• ANSWER : Call was answered
• CANCEL : Call attempt cancelled (i.e user hung up before the call connected)
• DONTCALL : Privacy manager don't call
• TORTURE : Privacy manager torture menu
• CONGESTION : Means Congestion, or anything else (some other error setting up the call)

Example 3: Dial multiple channels, partially delayed

Example 4: Answering machine (by dsfr aka Chris Hozian)

Example 5: Using Bristuff

Example 6: Using IAX2 friends

Using IAX2 friends with Asterisk@Home AMP

1. Ok, start up your AMP web portal and go into Setup (enter username and password as appropriate).
2. Go into your trunks and Add IAX2 Trunk. This trunk will NEVER be used in Outbound Routing. This trunk is merely a placeholder so it can be used by the custom trunk below.
   1. Don't worry about Outgoing Dial Rules for now, we'll worry about that later.
   2. In the Outgoing Settings section set the Trunk Name to value1.
   3. Put the contents of the [value1] heading (from above) in the PEER Details
   4. Change the line context=iaxhost2-in to context=from-internal
   5. Remove everything from the USER Details textbox.
   6. No other settings should need to be adjusted, so we'll Submit Changes now.
3. Now we need to Add Custom Trunk.
   1. Setup the Outgoing Dial Rules here. We're going to just enter 99205 & 982XX; each on their own line.
   2. Enter the Custom Dial String as IAX2/value3:value4@value1/$OUTNUM$
   3. (Optional) Setup any of the other fields that you didn't in the previous trunk now.
   4. Now we can Submit Changes to save this custom trunk.
4. Time to setup Outbound Routing.
   1. Enter iaxhost2-ext as the Route Name. Leave the Route Password blank.
   2. Enter the dial patterns from the Custom Trunk above to the Dial Patterns here. It may be redundant, but so what.
   3. Select the Custom Trunk above in the Trunk Sequence.
   4. Submit Changes
   5. (Optional) Adjust the routing priority with the Up/Down arrows in the top right (assuming you have more than 1 route).
5. You're all set on this server. Use the configs above for iaxhost2 and you are done.

• Asterisk@Home's AMP has a really nifty feature for Dial Patterns. Try out the Insert dropdown and select Lookup Local Prefixes. Now enter a local prefix in your area. You'll see all of the local dialing possibilities!

Return values in channel variables

• For PRI connections over a ZAP channel, as well as IAX2 channels, the hangup result code will be found the HANGUPCAUSE variable.
• Note that causecode 0 is effectively a notice that the causecode has not been set. Causecode 16 is not an error, but simply a notice that the call went through and was terminated normally.
• For all channels, the DIALSTATUS variable contains the result of the call, which may be used for dial plan logic
• The variables DIALEDTIME and ANSWEREDTIME contains timing for billing (as in CDR records)

Version comments

• Option A and S: added to CVS after release 0.7.2
• Option D: in CVS since May 2004
• Option f: added to CVS in June/July 2004
• Variables DIALSTATUS, ANSWEREDTIME and DIALEDTIME: added to CVS head in june 2004

Support for specific hardware

• Set variable ALERT_INFO to change ring cadence on Cisco 79xx phones. See also MySQL custom ringtones
• Phones running the SCCP (skinny) firmware have some support for pushing XML pages. If you want to test it, set the variable VXML_URL to point to a Cisco XML file on a web server.
• For distinctive ringing on Cisco Ata, see Asterisk phone cisco ATA18x

See also

• Asterisk Visual Dialplan, visual development platform for Asterisk dialplan development
• Asterisk cmd RetryDial
• Asterisk cmd FollowMe
• Asterisk auto-dial out
• For a Least Cost (and Failsafe) Dial Command see Application LCDial
• Configuration
• The Dialplan - extensions.conf