Dial() Bug


#1

I’m experimenting problems with Dial() function using this scenario.

Phone 1 Call ---------> phone 2 Answer (Phone 2, dial to extensions 0 to redirect the call to the receptionist)… (Phone 1 hear a total SILENCE…)

BUG: At the point phone2 dial to 0 and then transfer the call to the receptionist, the phone 1 can’t hear "Ring Back Tone"
the tipicall Ring… Ring… and i tested using MOH during the call redirection and it just keep in silent, so: No ring back tone, no moh works in that point.

Case: The main dialplan to call the receptionist was this:

exten=>0,1,Answer()
exten=>0,2,Dial(SIP/reception)

So, i try using this:
exten=>0,1,Answer()
exten=>0,2,Dial(SIP/reception,tr)

With this one too:
exten=>0,1,Answer()
exten=>0,2,Dial(SIP/reception,m)

Any of this dialplans fix the trouble… WTF! So, How i fix that?? [Simple =D]

I just add this to the dial plan:

exten=>0,1,Answer()
exten=>0,2,Dial(SIP/reception,1)
exten=>0,2,Dial(SIP/reception,21)

Now, when someone dial 0, it will call SIP/reception for 1 seconds, then call again using the timeout we want.
The 1st seconds is a fake that fix the situation, Now everybody’s fine :smiley:

Now looking back again, the problem was this

Phone 1 Call ---------> phone 2 Answer (Phone 2, dial to extensions 0 to redirect the call to the receptionist)… (Phone 1 hear a total SILENCE…)

Now the solution give this:

Phone 1 Call ---------> phone 2 Answer (Phone 2, dial to extensions 0 to redirect the call to the receptionist)… (Phone 1 hear a Ring…Ring…) [Or the music on hold if you decide to use the option m in the dial options]

Hope Asterisk developers compile this, for now my solution is totally working 100%.


#2

You shouldn’t be using Answer(), which will complicate the behaviour, as it will cause Asterisk to fake the ring back tone, rather than for the phone to generate it.

I don’t think you have identified the version of Asterisk.

You haven’t identified the model of hte phone.

For SIP transfers, it is highly desirable to provide SIP traces.

Finally, this is an Asterisk Support question.


#3

And, if there is a bug here, it won’t be in Dial().


#4

Asterisk-1.8.4.2
Linksys phones
When i was using asterisk-1.4.x the client who call was able to liste ring back tone


#5

Any answers?


#6

And the sip trace is where ???

Without a sip trace its impossible to say whats happening, I have never liked linksys hand sets as they use a odd method of transfer.

Ian


#7

This is what is happening.
Pepe call office --> FXO1 answer and call the ext. 201, phone 201 begin ring…ring… and a staff answer the call. Then the staff transfer the call to phone 202 using the phone feature "Xfer"
At this point the phone 202 is ring…ring… BUT Pepe just listen a SILENT, no ring…ring… to notify that pepe was transfered to another person (In this case, ext. 202)

What is the problem
here is the dialplan

[from-pstn]
exten=>s,1,Answer
exten=>s,2,Dial(SIP/201)

[from-internal]
exten=>_2XX,1,Dial(SIP/${EXTEN})


Actually i fix the problem using this dial plan, check carefully!

[from-pstn]
exten=>s,1,Answer
exten=>s,2,Dial(SIP/201)

[from-internal]
exten=>_2XX,1,Dial(SIP/${EXTEN},1)
exten=>_2XX,2,Dial(SIP/${EXTEN})

Using this dial plan, everything work perfect. But using older version of asterisk,dahdi,addons,dahdi-linux,etc… Using another old versions i never have this kind of issues.