Why can't I transfer?

I have a strange problem,
It's the same as described here: http://forums.asterisk.org/viewtopic.php?p=178420
But I haven't set up any custom stuff, All my extensions and queues are created in freepbx.
people can transfer between extensions if the call is internal, but if the call comes from a queue, then it gives that "not a valid extension" error as soon as you press anything after initiating the transfer - exactly as described in the link above.
here's my extensions file:
As I say, all the settings were done in freepbx.
I've got asterisknow 3 installed.
I did try yum update - and installed the updates, but that didn't help.
Who is Participating?
Again -- I would recommend PBXinaFlash Green.

Let me now if we can help in any way..
Looking at Distro watch as I had not seen this one before --

What version of AsteriskNow, Asterisk and FreePBX is the distro you built utilizing ?


StevenHookAuthor Commented:
Asterisknow 3, Asterisk 11.4.0, Free PBX framework (this is after yum update)

I have another system running asterisknow 2.something which is OK - kinda - except that after a yup update, it stopped working - I have to boot to an older kernel for asterisk to run.

I am looking for a system that's going to be upgradeable well into the future, I had trixbox, and that died horribly, I don't want to pick a distro that's headed for the grave again. I assumed that digium, as it makes all the interface cards we use with asterisk, would be the best to go with, they also have the DAHDI stuff built-in which helps a lot - we use PRI cards.

Network Scalability - Handle Complex Environments

Monitor your entire network from a single platform. Free 30 Day Trial Now!


  I would use the PBXinaFLash Distro or the FreePBX distro.  

  The general consensus in the community is never YUM update the kenral unless something is seriously broken or a major security flaw has been discovered.  It's usually better to do a new install from Distro and then restore / recover the parts like FreePBX or /tftpboot..

  I think I would look in /var/log/asterisk for the full.log and run a test and see why the transfer is failing --
StevenHookAuthor Commented:
If I watch the CLI - it says that that number isn't in the context.
[2013-06-28 08:26:48] WARNING[17258][C-0000000a]: features.c:2546 builtin_blindtransfer: Extension '32' does not exist in context 'macro-dial-one'

Open in new window

But that's really pushing the limit of what I know about asterisk.
I actually dialed "320" but it errors almost immediately - seems like the "0" doesn't even get a chance to get in there.
I've never played with any of the stuff like contexts - I leave the default stuff that freepbx does alone.
The queues and extensions were restored from a backup of another machine, I thought that might be a problem, so I created some test extensions and queues from scratch, but they behave the same.
I know you like PBXinaFlash, but I haven't heard much about it, why do you prefer it? I think I would feel safer with FreePBX, if I am going to need to choose a new distro.
Will I be able to restore my backup with all my extension and fax and queue settings?

    Right now PBXinaFlash is probably the largest and most widely used an it builds from current source.   Spend some time in the forums on both sites reading, or spend a weekend downloading and building both products.

    My issue with Digium is they have at least three versions now that compete against each other for attention and resources.  

    The log extract does not make sense.

     Not sure how to help more other than possibly doing a work session with you on you system.  ========
StevenHookAuthor Commented:
I think I will reinstall it.
I must just pick one. :)
StevenHookAuthor Commented:
Why doesn't PBXinaflash like 64 bit systems?
with all the simultaneous calls and recordings and MOH messages doesn't it want a lot of RAM?
StevenHookAuthor Commented:
Just as a side, I've noticed that some call centres have pre-recorded agent greetings.
Do you know of any plugins or modules that would let me make pre-recorded greeting and have them played to the caller and the agent when they pick up the phone?
There are 2 company names, so depending on the incoming route I'd like 2 different greetings?

-- 64Bit I believe,  although am not positive, is more an issue of the code base than the ISO.  

--  Yes when you setup your Queue just add an announce and then have you incoming route send the call to the proper queue based on the dialed number.

StevenHookAuthor Commented:
Isn't the announcement payed when joining the queue - or when the call is picked up?
like a "Hello, thank you for calling Yadda, my name is Yadda, how can I help you?"

So should I go with the 32bit piaf?
Missed the agent part --  

    Sorry, yes the Announce is on Join and/or every so many .......  when on hold..
Question has a verified solution.

Are you are experiencing a similar issue? Get a personalized answer when you ask a related question.

Have a better answer? Share it in a comment.

All Courses

From novice to tech pro — start learning today.