I have only this simple configuration, no SIP-trunks etc configured.
My problem:
SIP-softphone at extension 100 calls to queue at 200 and gets connected there as expected. Then I wish to transfer the caller from the queue to the conference at 300. I’m trying to do this with the Asterisk Java-API using the AsteriskChannel.redirect-method, but the caller gets disconnected every time.
What could be the problem? Should this be possible?
EDIT1: Running Asterisk v11.7.0, maybe this is a bug in this version?
EDIT2: On the Asterisk LOG I see text like this:
[2014-11-18 07:43:55] VERBOSE[5823][C-00000007] pbx.c: == Spawn extension (300, default, 1) exited non-zero on ‘SIP/100-00000007’
[2014-11-18 07:43:55] WARNING[5823][C-00000007] pbx.c: Channel ‘SIP/100-00000007’ sent to invalid extension but no invalid handler: context,exten,priority=300,default,1
So, is the conference extension 300 an invalid extension for transfer purposes?
Channelredirect bypasses any transfer restrictions. The error means that you have no lines similar to:
[default]
exten => 300,1,…
in your dialplan. Maybe you are transferring to the wrong context?
Incidentally, in Asterisk terms, you cannot have calls coming from extensions. extensions only refers to what is in extensions.conf. The source phone may be unreachable from there, may have multiple extension numbers, or may have different numbers depending on the calling context.
Ah, thanks for the hint. I’m total newbie with Asterisk so far so learning a lot of stuff here.
Took a look at the extensions-configuration and realized that the “default” context was wrong, when I changed it to “ext-meetme”, redirect worked fine!
Using FreePBX as well so I guess the one thing I didn’t realize that the conference numbers and extensions are generated under separate contexts, thus the “default” context wasn’t correct on moving the caller from queue to a conference.