* You are viewing Posts Tagged ‘transfer’

mISDN issues again

Hi
 
I been testing my clients installation for any problems with the farsouth
gateway and what I noticed is that the calls are being dropped at transfer.
After testing
I found that when the reception hits the transfer button on her sip phone, the
caller gets a dialtone and the reception goes on hold. Very wierd problem.
 
Ive disabled senddtmf and astdtmf in misdn.conf , that seems to have solved the
problem. Any insight on this.
 
 
Zakir

Blind transfer failed, SIP REFER Method

Hi,

I setup an asterisk system (version 1.8.0-rc5). While using a SIP only
environment I discovered a problem using blind transfer. The phones are
SNOM or Aastra and are using the SIP REFER Method.

The following is working:
User A calls user B, B accepts the call, user A than transfers to user C

The following is NOT working:
User A calls user B, B accepts the call, user B than transfers to user C

The call is terminated by asterisk without any warnings or error message
in the CLI.

Looking at Events on AMI, I can see in the first case an Event
“Newchannel” with a “Channel: AsyncGoto…” followed by an Event
“Masquerade” in prior to the “Transfer”. These events are missing in the
second case.

Is this a new bug or do I something wrong? Shall I open an issue on the
tracker?

Thanks for any hints,

Karsten

How to use Atxfer in AMI

Hi,

I’m trying to make a attended transfer through AMI. I though i could use
Atxfer, and it seems ok, but nothing happens.

And I can’t find any how-to or description on how to do this. What more
do I have to do to make this work?

In Asterisk Call Manager:

Action: Atxfer
Channel: SIP/36-xxxxxx
Exten: 33
Priority: 1
Context: Phone

Response: Success
Message: Atxfer successfully queued

Best regards,
Kent Varmedal

How to change features.conf’s atxfer dialing tone ?

Hi,

I’m facing the following request :
“When someone is starting an assisted transfer using Asterisk’s features
codes, he will ear a prompt saying “Transfer” and then a dialing inviting
him to dial the number he tries to reach.
This tone volume is qualified as a bit too load.”

Is it possible to change that and have a more delicate volume ?

A quick look inside features.conf doesn’t show evidence.

Regards

How to tell if there is a transfer from CDR?

Is there any way to know if a call was transferred from reading the
CDR? Any relation in fields like UNIQUEID? Something that can be
scripted to make a special report?