Incoming call when I'm on the phone, strange result

Asterisk Asterisk Support
1

Following is a snippet out of my extensions.conf file, which rings my desk phone when someone calls into the PBX and selects my extension.

Various symbolics mean:
ERICSGS = my Grandstream phone on my desk
ERICSHP = XLite running on my laptop
LABVOIP = an IP phone in the lab
CONFVOIP = an IP phone in the conference room

exten => ${ERICEXT},1,Wait,1
exten => ${ERICEXT},2,Dial(${ERICSGS},30,t)
exten => ${ERICEXT},n,Goto(${ERICEXT},110)
exten => ${ERICEXT},103,Goto(${ERICEXT},110)
exten => ${ERICEXT},110,Wait,1
exten => ${ERICEXT},n,VoiceMail(u${ERICEXT}@default)
exten => ${ERICEXT},n,Goto(t,1)

If I’m on the phone and someone calls in and selects my extension, they go direct to voicemail. So far so good.

Now, instead of the above, suppose I have:

exten => ${ERICEXT},1,Wait,1
exten => ${ERICEXT},2,Dial(${ERICSGS}&${ERICSHP}&${LABVOIP}&${CONFVOIP},30,t)
exten => ${ERICEXT},n,Goto(${ERICEXT},110)
exten => ${ERICEXT},103,Goto(${ERICEXT},110)
exten => ${ERICEXT},110,Wait,1
exten => ${ERICEXT},n,VoiceMail(u${ERICEXT}@default)
exten => ${ERICEXT},n,Goto(t,1)

The idea is that I’ll get the call whether I’m at my desk, in the lab, in the conference room, or somewhere else running XLite on my laptop.

That all works pretty well UNLESS I’m on the phone when the call comes in. Then, instead of the call going directly to voicemail like it should, it rings all the other phones in the Dial group. The caller hears nothing … no ringing signal or anything … for 30 seconds then he gets directed to voicemail.

I don’t think this is how it’s supposed to work … the explanation of the Dial(${1}&${2}&…) that I read says that the first one to answer satisfies the Dial command. That would seem to imply that if ANY ONE of the phones in the group is off-hook when the call comes in, it should register as a busy, and redirect the call to voicemail.

Right?

In any case, it ain’t working that way. Anything I can do to fix it, or do I misunderstand how it’s supposed to work?

2

Hi,

you read it the wrong way round:
if at least one phone in a dial command is available, it will ring. Only if all of them are busy, the call ends up as busy.
An agi script called from the dialplan could check if all phones are available, and ring them all together, or otherwise go the busy path

Musicman

3

if at least one phone in a dial command is available, it will ring.
Only if all of them are busy, the call ends up as busy.

Oh. :frowning:

Oh well, so much for that idea…

That raises to the forefront the other part of the question … why, if one of the phones is off hook and the others are rining, doesn’t the caller get a ringing tone? Incoming caller gets silence, until the 30 seconds times out then he gets voicemail.

I’ll look at the agi script option.

4

Hi,

I really have no idea about the tones - in fact I have recently posted a question about another situation where tones are missing

Musicman

5

Where you have the t on the end of the dial command, add r
eg Dial(sip/phone,20,tr)

Look up the options for the Dial command in http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial it wont do it unless you tell it too.

Mike.