Call Forker

Revision as of 16:28, 14 May 2013

This module is used to route a call to multiple alternative targets. The first target that answers will be connected with the caller while the other calls are abandoned.

A call fork is implemented by connecting the incoming call to a fork master. For each target a fork slave is created that generates a call.execute message to create the outgoing call leg. Once one of the outgoing calls answers all other active slaves are disconnected with a reason of "pickup" and the incoming call leg is connected directly to the answered outgoing one, finishing the forking process.

Note: Verify that the module is launched, otherwise add "callfork.yate=yes" to yate.conf.

Parameters for each fork branch

NOTE: Starting with Yate 3.3.3 (SVN Rev 4485 - 2011-07-12) the module supports separate parameters for each fork branch.

• The callto parameter of the call.execute message must be 'fork'.
  
• Each fork target must be specified in a callto.index parameter.
  
• Fork target parameters can be specified in callto.index.param_name parameters.
  
• The module will start looking for targets and parameters starting with index=1 and will stop when the parameter with current index is not found (e.g. for callto.1=sip/sip:1@host1, callto.2=sip/sip:2@host1, callto.4=sip/sip:3@host1 the module will handle the first 2 targets and will ignore callto.4).
  
• Special targets like | or |exec can be specified (e.g. callto.2=|exec).
  
• If the message is sent from a C++ module the callto.index parameter may be a NamedPointer carrying a NamedList with target parameters not prefixed (e.g. instead of setting callto.1.maxcall=10000 in the call.execute message a maxcall=10000 parameter can be set in the list of target parameters).
  

Examples

1. To use as a fallback routing module you should add in regexroute or your routing module something like:

   fork sip/sip:1@host1 | h323/2@host2 | sip/sip:3@host3;stoperror=busy 

2. To use call forking from a routing module you should return a string like: fork target1 target2 target3 ...

   fork sip/sip:1@host1  h323/2@host2  sip/sip:3@host3;stoperror=busy

3. Implementation in regexroute.conf:

^.*$=fork sip/sip:\0@host1 | h323/\0@host2 | sip/sip:\0@host3;stoperror=busy 

^.*$=fork;callto.1=sip/sip:\0@host1;callto.1.maxcall=5000;callto.2=|;
 callto.3=h323/\0@host2;callto.3.maxcall=7000;
 callto.4=|;callto.5=sip/sip:\0@host3;callto.5.maxcall=9000;stoperror=busy

 ^.*$=fork sip/sip:\0@host1  h323/\0@host2  sip/sip:\0@host3;stoperror=busy

 ^.*$=fork;callto.1=sip/sip:\0@host1;callto.1.maxcall=5000;callto.2=h323/\0@host2;
 callto.2.maxcall=7000;callto.3=sip/sip:\0@host3;callto.3.maxcall=9000;stoperror=busy

Targets must be separated by spaces when set as a single callto (like 'fork sip/sip:1@host1 h323/2@host2').
A target with the name "|" (the pipe character) has a special meaning, it delimitates group of targets.
Groups are called in sequence with targets in each group called simultaneously.
Simultaneous targets result in having more than one fork slave at the same time.

Timed advance

Starting with Yate 2.2 it is possible to proceed to the next group of targets based on a timer. This is accomplished using a special separator target that still starts with a pipe character:

• |next=N - advance to the next group of targets after N milliseconds, keep calling current targets
• |drop=N - drop current targets and advance to next group after N milliseconds

If you have something like

fork sip/sip:1@host1 |next=3000 sip/sip:2@host2 

will start calling initailly 1@host1 and after 3 seconds of no answer will also start calling 2@host2

Jump out of a fork

Starting with Yate 3 (SVN Rev 3310 - 2010-05-11) is it possible to jump out of a fork and continue the call without the fork master.

• |exec - for next targets do not create new slave call legs but dispatch directly call.execute
• |exec=N - effectively identical to |drop=N |exec

Once the |exec target is reached all remaining targets will be attempted sequentially until call.execute returns true.

Example

fork sip/sip:1@host1 sip/sip:2@host2 |exec=20000 queue/desk wave/play/queue_full.au 

If the SIP targets do not answer in 20 seconds the fork is terminated and will try to send the incoming call to queue/desk. If that queue does not exist or the call cannot be queued it will end up connected to an announcement.

Parameters of the call.execute

The following parameters of the call.execute message are understood by this module:

• fork.stop - regular expression matching the names of the errors that will terminate the entire call. NOTE: this parameter was previously named stoperror - that form is OBSOLETE
• fork.fake - a media source that should be attached as fake early media while. The value whould be acceptable in as source in a chan.attach message
• rtpstrict - set this parameters to "true" or "yes" to drop with a "nomedia" error calls that don't acknowledge a RTP forward offer

IMPORTANT: All parameters apply to all branches of a forked call. You cannot set parameters specific to one target. Starting with Yate 3.3.3 fork parameters can be specified for each branch. For example a global fork.stop 'busy' can be overridden for a second target like this:

^.*$=fork;fork.stop=busy;callto.3.fork.stop=rejected;callto.1=sip/sip:\0@host1;
callto.2=|;callto.3=h323/\0@host2;callto.4=|;callto.5=sip/sip:\0@host3 

Starting with SVN Rev. 4527 (2011-07-28) the fork.stop condition can be reversed by adding a ^ character at the end. For example ^congestion$^ will terminate the fork on all errors except congestion.

The following parameters are understood in return from the call.execute message and apply only to that branch:

• fork.ringer - set this to "true" to have early media forwarded from that branch. NOTE: you should only apply this to the first target of a group as concurrency issues may appear otherwise
• fork.autoring - set this to "true" to have a call.ringing autogenerated for the call leg (but see below). This implies the same actions of setting "fork.ringer" to true
• fork.automessage - this provides a replacement for the message sent by fork.autoring and makes sense only if that is set. The default value is "call.ringing" and you may set it to "call.progress" instead.
• fork.calltype - this parameter controls the behaviour of the outgoing call leg. Set it to "auxiliar" to not be counted and be dropped when deciding to move to the next group, set to "persistent" to not be counted but persist until the end of fork

Other parameters (except "callto") are forwarded to the outgoing calls.

Configuration

There is no configuration file for this module. All parameters are taken from the call.execute message.

Example

A standard line for regexroute.conf will look like:

^1$=fork sip/sip:1@host1 h323/2@host2 | sip/sip:3@host3;stoperror=busy

That means when someone is calling extension 1 the call will proceed as follows:

• simultaneously call 1@host1 on SIP and 2@host2 on H.323
• if one call is answered connect to it and drop the other
• if a busy is received drop the entire call
• after both outgoing calls fail call 3@host3 on SIP
• if that fails too, finish the call with the most recent error (that returned by 3@host3 in the last step)

A demo query to return same from a database would be:

SELECT 'fork sip/sip:1@host1 h323/2@host2 | sip/sip:3@host3' AS 
location, 'busy' AS stoperror

(assuming you have result=location in register.conf)

Call forking when additional parameters have to be supplied

NOTE: This section applies for Yate older then 3.3.3 (SVN Rev 4485 - 2011-07-12)

Unfortunately routing a call to more than one target doesn't work if per-target parameters must be set. This excludes fork routing to registered accounts as they each need a different "line" parameter.

A workaround would be to catch and alter each individual call.execute message like this:

[default]
^.*$=fork $(rotate,$idx001, sip/\0@fwd, sip/\0@other, ...)
[extra]
call.execute=80
[call.execute]
${callto}^\([a-zA-Z0-9]\+\)/\([0-9]\+\)@\([a-zA-Z0-9_-]\+\)$=return;callto=\1/\2;
line=\3

This will turn any proto/user@acct into proto/user and set line=acct. The account name must contain only letters, digits, underscores and dashes. In particular a dot cannot be allowed as it would make impossible to discriminate regular domain names or IP addresses.

To apply more complex changes you can jump to computed contexts like:

[call.execute]
${callto}^\([a-zA-Z0-9]\+\)/\([0-9]\+\)@\([a-zA-Z_-]\+\)$=goto exec_\3;
callto=\1/\2;line=\3
[exec_fwd]
.*=return;caller=xxx
[exec_other]
.*=return;caller=yyy;callername=It's me!

For round-robin with fallback you would need something like:

^.*$=fork $(rotate,$idx001,| sip/\0@fwd,| sip/\0@other,| ...)

You should consider adding a maxcall parameter (maximum time until answer in milliseconds) so that unresponsive gateways can be skipped.

For round-robin without fallback you would need something like:

^.*$=$(index,$idx001,sip/\0@fwd,sip/\0@other,...)

Please see Routing for more examples of round-robin and/or fallback routing.

Forking a call routed by a different module

Sometimes you have to modify a call that is routed by another module like regfile. This is impossible using the call.route message but there is the option of altering the parameters of the call.execute message.

An example of implementing dropping call into voicemail would be:

[extra]
call.execute=80
[call.execute]
; can't fork an already forked call or calls to scripts (IVR and such)
${callto}^fork=return
${callto}^external=return
; but fork all other calls to the original and the leave mail script (after 30s)
${maxcall}^$=return;callto=fork ${callto} | external/nodata/leavemail.php;maxcall=30000

If the original called resource doesn't answer in 30 seconds the call will be routed to voicemail.

Providing fake ringback tone while fallback calling through different gateways

^1$=fork earlymedia sip/sip:1@host1 h323/2@host2 | sip/sip:3@host3;
stoperror=busy;maxcall=20000
[extra]
call.execute=80
[call.execute]
${callto}^earlymedia$=return;callto=tone/ring;fork.calltype=persistent;
fork.autoring=true;fork.automessage=call.progress

Starting with Yate 3.3.3 (SVN Rev 4485 - 2011-07-12) this can be set in a single rule:

^1$=fork;callto.1=tone/ring;callto.2=sip/sip:1@host1;callto.3=h323/2@host2;callto.4=|;
callto.5=sip/sip:3@host3;stoperror=busy;maxcall=20000;
callto.1.fork.calltype=persistent;callto.1.fork.autoring=true;
callto.1.fork.automessage=call.progress

The persistent tone/ring will not prevent moving over to call sip/sip:3@host3 but will persist and provide fake early media until the call is answered or finally fails.

Providing just ringback tone

Another solution is to use the fork.fake to set the fake early media. This is an easy way to provide custom ringback.

^1$=fork sip/sip:1@host1;fork.fake=tone/ring

See also

• Example of forking a call using lateroute
• Call End Causes