Summary:ASTERISK-08848: Can't transfer direct to Voicemail
Reporter:Kevin Savoy (ksavoy)Labels:
Date Opened:2007-02-19 21:49:31.000-0600Date Closed:2007-03-06 09:36:51.000-0600
Versions:Frequency of
Environment:Attachments:( 0) CLI.txt
( 1) extensions.conf
( 2) sip.conf
( 3) verbosedebug.txt
( 4) verbosedebug_with_sip_debug.txt
( 5) voicemail.conf
( 6) zapata.conf
( 7) zaptel.conf
Description:Trying to transfer to a voicemail box does not succeed.
Dialplan has _*40XX,1,(VoicemailMain,u)

When transferring the call goes to the voicemail and the message is heard but when pressing Transfer nothing happens on the phone.

The CLI displays the message "Transfer to an owned channel?"


I have tried this on Asterisk and it works. In 1.4.0 and the latest SVN it does not work. Something got broken in the new version.
Comments:By: Kevin Savoy (ksavoy) 2007-02-19 21:56:19.000-0600

Sorry to update I am using Fedora Core 4 with all the latest patches. I also tried the latest kernel with no luck.

The dial command is actually


By: Tilghman Lesher (tilghman) 2007-02-19 22:17:25.000-0600

That's probably because you don't have a mailbox that is *4000.  Your mailbox number is probably 4000.

By: Joshua C. Colp (jcolp) 2007-02-20 13:58:54.000-0600

I have been unable to reproduce this on the latest 1.4 from SVN, please give it a go and report back.

By: Kevin Savoy (ksavoy) 2007-02-21 15:29:11.000-0600

Ok I built up a completely new system using nothing but svn releases.
According to the CLI show version command shows:

Asterisk SVN-branch-1.4-r55914 built by root @ testpbx.novo1nd.com on a i686 running Linux on 2007-02-21 20:20:05 UTC

I used svn for asterisk-addons, libpri and Zaptel as well.

I still can't transfer direct to voicemail. I call into phone 4033. Press Transfer and then dial *4023. I then here my message that I recorded in the voicemail system so it is accessing directly to the voicemail. When I press Transfer to send the call to that voicemail nothing happens. I continue to hear the message.

I have attached my config files as well as the CLI screen showing the two attempts to transfer the call.

If you need anything else from me please let me know.

By: Joshua C. Colp (jcolp) 2007-02-21 15:45:12.000-0600

I need to know the steps you are doing with the phones. I have tried to recreate it as best I could from logs but still unable to reproduce. Correct any of these:

1. Phone 1 calls Phone 2
2. While Phone 2 is ringing Phone 1 puts it on hold and begins transfer
3. Phone 2 answers
4. Phone 2 puts on hold
5. Phone 1 transfers to voicemail, hears it, attempts to complete transfer.

By: Kevin Savoy (ksavoy) 2007-02-21 15:53:20.000-0600

No I use three phones.

Phone 1 my cell phone calls phone 2 extension 4033.
Phone 2 4033 tries to transfer phone 1 to voicemail of extension 4023.
Phone 1 hears hold music
Voicemail starts to play voicemail message of 4023 on phone 2 ext 4033.
Phone 2 while hearing voicemail message presses Transfer to complete the transfer of phone 1 to voicemail of 4023.

Transfer should complete and phone 2 should drop out.

Instead phone 2 continues to hear voicemail message and phone 1 hears nothing.

I have also tried this with a blind transfer.

I have built 4 machines from scratch now and all do this. Not sure why you can't duplicate.

I can send you my install instructions as to how I installed from scratch to finish if you'd like. Do you have access to Fedora Core 4?

By: Joshua C. Colp (jcolp) 2007-02-21 16:03:08.000-0600

Ah it is just in your CLI output it's SIP/4033 that is calling itself... so I was slightly confused.

By: Serge Vecher (serge-v) 2007-02-21 16:03:31.000-0600

alright, since you are using SIP for the transfering phone, let?s see the sip debug as per following:

1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterisk with the following command:
  'asterisk -Tvvvvvdddddngc | tee /tmp/verbosedebug.txt'
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Reproduce the problem
6) Trim startup information and attach verbosedebug.txt to the issue.

By: Kevin Savoy (ksavoy) 2007-02-21 16:58:45.000-0600

Hope I got this right and that it helps. log is uploaded

By: Joshua C. Colp (jcolp) 2007-02-21 18:46:02.000-0600

I do not see any transfer attempts in the latest verbose debug. These come up as REFER packets.

By: Kevin Savoy (ksavoy) 2007-02-21 21:37:22.000-0600

That's what I'm trying to tell you. Nothing happens.

The system doesn't even see the fact that the transfer button has been pushed.

However this DOES work with and not 1.4.

By: Serge Vecher (serge-v) 2007-02-22 09:03:09.000-0600

ok, please get us the same output from so we can see how and why it works there. Thanks for helping to track this down.

By: Kevin Savoy (ksavoy) 2007-02-22 15:10:17.000-0600

I am out of town right now and won't be able to do this until at least Monday or Tuesday

By: Joshua C. Colp (jcolp) 2007-02-26 14:50:04.000-0600

Just to clarify a bit more... the transfer attempt never came in from the device so there might be some signalling issues before hand but once I see the 1.2 debug I can compare.

By: Kevin Savoy (ksavoy) 2007-02-28 14:04:52.000-0600

Ok I've uploaded verbosedebug Successful Transfer.txt which was made on a newly rebuild test machine with on it.

I do not have an outside line on that box so I dialed from 4023 to 4033 and then transferred the call to *4023 and it worked.

By: Serge Vecher (serge-v) 2007-02-28 14:17:38.000-0600

you've forgotten to enter 'sip debug'

By: Kevin Savoy (ksavoy) 2007-02-28 14:36:05.000-0600

Grrr..stupid me. Ok this one has the debug turned on I hope :)

By: Serge Vecher (serge-v) 2007-02-28 15:01:41.000-0600

interestingly enough, the phone does send a REFER when is used to do the transfer. Why doesn't it try to transfer with REFER in the case of 1.4? It is listed in the Allow field ...

By: Kevin Savoy (ksavoy) 2007-03-01 12:45:13.000-0600

Are you asking me the above question or was it retorical? :)

If you're asking me, that's why I'm asking you! :)

By: Kevin Savoy (ksavoy) 2007-03-06 09:34:13.000-0600

Interesting. I thought I'd check to see if the new release 1.4.1 would have fixed this problem and it did!!!

Apparently someone fixed it without even knowing it was a problem.

Well it works now and that's the main thing.

Thank you for your assistance with this and we can consider this issue closed.

By: Serge Vecher (serge-v) 2007-03-06 09:36:50.000-0600

cool, thanks for reporting back.