abefroman at January 12th, 2009 16:35 — #1
How do I transfer a call to another extension?
I'm talking to a customer, he wants to speak to another guy, what do I have to do to send that call to the other guys phone?
fskrotzki at January 12th, 2009 17:01 — #2
there are several answers to this question.
First off go to the feature codes and make sure that you have "In-Call Asterisk Attended Transfer" and "In-Call Asterisk Blind Transfer transfer" enabled and check what the key codes are.
Next depending on your phone (and this is a a BIG depends). try and dial that pattern. If you are using a ATA it is very possible that it is blocking the key sequence and you'll either need to unblock that pattern or change the pattern.
Also depending on your phone it might have a transfer button that can be programmed to do the transfer making it easier.
What I can tell you is google is your friend. Providing more details like phones, etc. helps. If I had asked how to change the oil on my car you'd probably say it's easy also, but without knowing the exact details like make and model you could not give a person who know nothing about cars a good set of instructions either.
abefroman at January 12th, 2009 17:23 — #3
Its *2 withouth pressing hold, so the caller will hear the beeps, then when I am at the dial tone, I put in the extension I want to transfer it to.
I have an Aastra 480i phone and don't see a transfer button, if I had that, then would the caller not hear the beeps for *2?
wiseoldowl at January 12th, 2009 18:02 — #4
One small nit to pick with fskrotzki's comment - while an ATA might pick off certain feature codes during initial dialing, once a call is in progress it should not respond to anything dialed by the user (one possible exception is **** when dialed on a Linksys/Sipura device). Of course, I can't say with absolute certainty that there is no poorly designed unit that won't interfere, so YMMV.
The second issues with a device or SIP-based phone is whether you want the adapter to handle the transfer, or Asterisk itself. I think in many cases the net effect is the same, but it may mean there are two different dialing patterns that will work. Normally, if you want the adapter or phone to do initiate it, you flash and dial a code, or press a transfer button if one exists. I believe this then sends some kind of SIP message back to Asterisk telling it to reroute the call, but don't quote me on that. The other way is to directly interact with Asterisk, which is accomplished by pressing ## (for a blind transfer) or 2 (for an attended transfer) during an in-progress call (I notice you have to dial the two characters within a fairly short time, or it doesn't "take" - 2 is especially hard to dial quickly enough - also these codes can be changed on the Feature Code Admin page). You will then hear Allison say "Transfer" followed by a dial tone, after which you dial the extension or number you wish to transfer the call to. My personal preference is to use Asterisk's transfer feature (since I know it definitely pulls the call back from the phone) but as I say, that may happen anyway even if you use the transfer feature built into the phone or device (although you probably won't hear Allison).
Feel free to correct me if I am wrong about any of this!
fskrotzki at January 13th, 2009 09:52 — #5
I would be shocked if the 480i does not have a transfer button.
one minute later...
Don't take this the wrong way but you need to learn to use the internet a bit better. Yes all transfer modes are supported on the Aastra 480i, it's even covered in the user manual on page 19. most of the Aastra phones are pre-configured at the factory to support Asterisk right out of the bx you just need to assign extension, id, password and server. We have 40+ 9133i phones.
wiseoldowl is correct most ATA's are not a problem but there are one or two that with the factory default dial patterns and settings do block one of the two transfer modes. It's been dicussed on this forum a few times.
sri2talk at March 28th, 2009 12:24 — #6
I've configured ATA to send DTMF through RTP (RFC2833).
I diall ##, I get "Transfer" prompt only at times. The ## should be pressed in very quick succession. *2 does not work at all. The Feature is enabled on FreePBX.
How can I tune the inter-digit timeout for blind transfer for the SIP ATAs?
- Should it be done at the ATA?
- Is there a param on the FreePBX/asterisk side?
fskrotzki at March 30th, 2009 09:20 — #7
Let's see you want help but can't even tell us which of the hundreds of ATA's out on the market you are using.
So about the only help I can give you is the following:
Contact the ATA's manufacture, get the administration manual for the unit and read it. Yes it's blocking the code due to it's internal processing before sending it off.
sri2talk at March 31st, 2009 21:22 — #8
Sorry about keeping the ATA brand a secret. I'm using Grandstream 486.
I've configured the ATA to send DTMF through RTP (rfc2833). I've disable call features on the ATA.
I've configured the SIP "dtmfmode" as "auto" in freepbx for this extension.
I tried again and it works sometimes. What I observed was that the ## should be pressed in quick succession. How do I set the inter digit timeout so that there is some tolerance to dialing the second #? *2 works sometimes.
*2; if not dialing begins within 5 seconds, the parties A and B are connected back again.
Where do I tune the "5 seconds" ?
raphetvargas at February 19th, 2014 16:39 — #9
What if I have directmedia enabled and so therefore asterisk will not recognize codes? Is there any other way to do trasfers? Blind or Attended?
The endpoint should support this, however, I have tried with Zoiper and eyeBeam (x-lite) and it will not work.
system at June 4th, 2014 14:46 — #10
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