Call forward after x seconds , and both ring (no queue/follow me)

Hey, is it possible to have a dialplan that forwards the call after x seconds, butall phones should still ring

Lets say , 100 is first phone, 101 + 102 + 103 are backup phones
100 starts ringing, after 10 seconds 100 AND 101 +102+103 are all ringing?

without the use of queue/ring groups ?

thnx in advance

same => n,Dial(PJSIP/100&Local/101@DelayCall&Local/102@DelayCallLocal/103@DelayCall)

[DelayCall]
exten => _X!,Wait(10)
same => n,Dial(pjsip/${EXTEN})
1 Like

What’s a ring group? Asterisk doesn’t use that term.

hey @TheMark , thnx for feedback, diaplans is something new for me,

i aready had 1 line in my diaplan, to make sip calls between extensions:

exten => _X!,1,Dial(SIP/${EXTEN})

so i placed your code below it , converted to SIP (pjsip) is coming
so my extensiosn file looks like :

exten => _X!,1,Dial(SIP/${EXTEN})

same => n,Dial(SIP/8002&Local/100@DelayCall&Local/101@DelayCall)
[DelayCall]
exten => _X!,Wait(10)
same => n,Dial(SIP/${EXTEN})

problem 1
Now,offcourse if i call from 8003 to 8002 , then it dials 2 times :slight_smile:
first because of default diaplan, second, i think because of your added lines…

prpoblem2 , if i can fix problem 1 , i see error below?

is there something wrong with that “local” stuff?

[Jan 17 08:38:46]     -- Executing [8002@default:2] Dial("SIP/8003-00000000", "SIP/8002&Local/100@DelayCall&Local/101@DelayCallLocal/102@DelayCall") in new stack
[Jan 17 08:38:46]   == Using SIP VIDEO CoS mark 6
[Jan 17 08:38:46]   == Using SIP RTP CoS mark 5
[Jan 17 08:38:46]     -- Called SIP/8002
[Jan 17 08:38:46] NOTICE[499][C-00000001]: core_local.c:739 local_call: No such extension/context 100@DelayCall while calling Local channel
[Jan 17 08:38:46]     -- Couldn't call Local/100@DelayCall
[Jan 17 08:38:46] NOTICE[499][C-00000001]: core_local.c:739 local_call: No such extension/context 101@DelayCallLocal while calling Local channel
[Jan 17 08:38:46]     -- Couldn't call Local/101@DelayCallLocal/102@DelayCall
[Jan 17 08:38:46]     -- SIP/8002-00000002 is making progress passing it to SIP/8003-00000000

The provided code replaced your original Dial.

There is a priority missing on the first line of the DelayCall context (I would also suggest “.” is better than “!”).

ah ok, but if i overwrite that line, what about my other sip calls? do they still work?

I’ve no idea what your other SIP calls are, so I can’t answer that.

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