Call Transfer Trouble

General Help
1

First off Happy New Year to all…

I figured I would throw this out and see if anyone has any suggestions. I am having a problem where if someone tries to transfer a call to a different extension, the call just gets dropped.

My configuration is that I am running Asterisk 1.6.2.15 with FreePBX 2.8.0.4, and I have a mix of Aastra 6739i and 6731i phones.

If I take and answer a call, and I want to transfer it to a different phone, if I hit the transfer key, it puts the caller on hold (they get music on hold), and it prompts me for an extension. I then enter the extension to transfer to, and hit the button to perform the transfer, and the call is just dropped.

Here is a log snippet:

[Dec 31 11:41:40] VERBOSE[7457] res_musiconhold.c:     -- Started music on hold, class 'default', on SIP/bandwidth_east-000000bb
[Dec 31 11:41:44] VERBOSE[7457] res_musiconhold.c:     -- Stopped music on hold on SIP/bandwidth_east-000000bb
[Dec 31 11:41:44] VERBOSE[7457] pbx.c:     -- Executing [h@from-internal-xfer:1] Macro("SIP/bandwidth_east-000000bb", "hangupcall") in new stack

No errors, no transfer, it just hangs up. So not sure if this is a phone issue, or what, but seems strange a modern Aastra would have a transfer issue. I sure couldn’t find anything googling around.

On the other hand, if I just park the call, not transfer it, then it works fine.

Any ideas what could be wrong??

-Howard

2

Guys,

It was mentioned elsewhere in the forum but just make sure you are not being bitten by the SIP REFER (blind transfer) / manager transfer bug that affects Asterisk 1.4.38 and 1.6.15. You mention you are running these versions–seems likely to be the source of your trouble.

https://issues.asterisk.org/view.php?id=18185

Get on 1.6.2.16-rc1 or 1.4.39-rc1 at least.

3

I had the same problem on my 31is using firmware 2.6.0.1007

argg - spoke too soon, this appears to only happen on inbound calls. If I make an outbound call, then try to transfer it to another extension it works fine.

I am using Asterisk 1.4.38 and freepbx 2.8.0.4 with latest modules as of today.

4

Here is a full sip debug if anyone is interested in debugging
http://pastebin.com/fTRfDqXa

5

What happens when you do a ## or *2 transfer?

Bill