Unable to route to Sip and POTS line at the same time


#1

Here’s my scenario,

I receive an outside call. Once this outside call reaches *, i wish to dial Sip devices as well as another outside line (i.e. my cell phone). When I first receive the call, my Sip devices ring, but once a bridge is made between my incomming call and my other outside line (i.e. my cell phone), my Sip phones get kicked off. As well, timeout for my dial command no longer works either.

Help is appreciated.


#2

That actually sounds normal.

Were you hoping that a call directed to multiple points, and answered at one of the points, would continue to ring the remaining points?

What were you hoping might be able to happen if one of the secondary points answered the call? Would you want it to pull the call away from the first answering station? Would you want all three points joined in a conference?

Most systems release control of a call to the first point that answers it. I’d expect that Asterisk would do the same.


#3

To your first question. No.

I actually don’t have any point answering yet. I’m trying to direct a call to multiple points but one of my points is an outside line. And once the call gets directed to an outside line (but not answered), all other points somehow get terminated.

i.e. I’m using something like the following command

exten => s,1,Dial(Sip/John&Sip/Jill&Zap/g1/5555555,15,t)

As soon as * tried dialing the “5555555” number, all other points get dropped.

Hope this makes more sense.


#4

Ok… I noticed that your timeout is only 15 seconds…

That’s only three rings, and cell phones are notoriously slow to connect. By the time your cell starts to ring, your timeout may occur and make all the other sets stop ringing.

Try increasing that time first. That may fix your trouble.


#5

My actual timeout was for 50 seconds as opposed to 15 seconds…my SIP phones ring once and then get dropped as soon as a connection is made to my outside line.


#6

contextually speaking, it still seems to be normal behavior as you are combining a local physical context with an off-premises context in a most atypical fashion. The default behavior is to omit that which is disparate from where the call is sent, so it seems normal (by default) that the local extensions are pulled out of the loop when the call is sent somewhere in the cloud (which is no longer a local physical context). Now, that’s not to say that you can’t configure the bridge-group to NOT automatically occur, and i’m not sure how you would go about that; but as far as the DEFAULT behavior, it is most normal from what i can tell spatially speaking. I’m thinking the best way around this is to setup a callgroup or a queue. I would be surprised if the native bridging behavior, that is to bridge the disparate source and destination targets, omitting any other physical prem, is very configurable. callgroup, pickupgroup etc seems like something more suited to handle a physically disparate logical grouping of phones since they are not all local. This may be one you want to ask in the dev forum…


#7

I have the EXACT same problem.

I have just tried for 2 hours to have a caller ring both an IAX phone (or SIP phone) and an external phone (my cell phone) SIMULTANEOUSLY until the first phone picks up.

I have tried it as

  1. ring group to IAX and external number
  2. ring group to IAX and local extension that is fowarded to external number
  3. call queue to IAX and external number
  4. call queue to IAX and local extension that is forwarded to external number
  5. local extension with Dial to IAX and ZAP number
  6. local extension with Dial to IAX and local extension that is forwarded to external number

In ALL of these cases, the IAX (or SIP) extension rings once then immediately stops ringing once the cell phone starts ringing. Picking up the IAX or SIP phone after it stops ringing yields a dialtone and the cell continues to ring.

Has anyone gotten this setup to work?


#8

I have this same setup, but mine works fine. I ring multiple sip telephones, and multiple outside lines. I use the confirm tag on the outside lines, though. I did it keep all my voicemail in one place.

Here is the line I use:

Dial(SIP/xxx&SIP/xxx&${INSIDETRUNK}c/xxx&${INSIDETRUNK}c/xxx&${TRUNK}c/xxxxxxx,30,tm)

where the x’s are numbers dialed.

Joseph “Darin” Thomas Sr.


#9

in your logs do you see that the Zap channel is connected way before the cell phone rings ? as far as Asterisk is concerned, Zap “answers” as soon as you’ve finished dialling :frowning:

it’s far from ideal. have a search around the forum, people have posted solutions that will do what you want, but the dialplan is a bit more complicated. you probably want a solution that makes the call recipient press a key to accept a call, that way the caller doesn’t get bridged to a telco announcement or your voicemail.


#10

That is what the little c tags on the zap extensions due, it forces the person answering the zap call to confirm answer by pressing the pound key. That is why my setup is working.

Dial(SIP/xxx&SIP/xxx&${INSIDETRUNK}c/xxx&${INSIDETRUNK}c/xxx&${TRUNK}c/xxxxxxx,30,tm)

Joseph “Darin” Thomas Sr.