[Asterisk-Users] Dropping incompatible voice frame

[Kevin Savoy]

This didn't work for me either. I tried using the patch at the link below
and it didn't work either. 

 

If I were to guess what was happening here, it would be when the call is
forwarded by the phone Asterisk doesn't know which device to send the call
to. How does it know to open a Zap channel and dial the command? What tells
Asterisk to open Zap channel and dial the number the phone had it it's
forward? Am I off track here?

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joe Pukepail
Sent: Wednesday, June 28, 2006 10:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dropping incompatible voice frame

 

This is  known issue, we fixed it by putting an answer() in the dial plan
before it gets forwarded, the fix  transcode_via_sln=no (detailed in the bug
tracker) didn't work for me. YMMV. 

 


http://bugs.digium.com/view.php?id=4101
 

On 6/28/06, Kevin Savoy <ksavoy at novo1.com> wrote: 

Sorry if this has been posted before but I'm having an issue where I get the
following on my CLI.

 

ast_read: Dropping incompatible voice frame on Local/XXXXXXXXXX of format
ulaw since our native form has changed to slin

 

A call comes in on our main to toll free number on an AT&T T1 line and is
sent to phone 4000. This is our secretary's desk. If she leaves the desk she
forwards the phone to one of our sister companies so that they would answer
the call. This call is sent back out the AT&T T1. If she answers the call
and then forwards outside the building it works fine but if she forwards her
phone outside the building to auto forward the call when she is away from
her desk we get the above error. I have recreated this on my own phone (both
hers and mine are Polycom 501's) and with a Cisco 7960. I also tried a
different toll free number with the same results. I searched the internet
and found four people having the same issue but none have gotten responses
on how to fix it. Each time it was something similar where the call was
redirected. I know the T1's are configured correctly because all other
incoming and outgoing calls work fine until this error occurs. Then nothing
works. 

 

I am using Asterisk 1.2.7.1 <http://1.2.7.1/>  with Zaptel 1.2.5 and Libpri
1.2.2 . I have tried using both a digium Wctxxp 4 port and RedFone's
Fonebridges and have gotten the same result both ways so the problem is
within Asterisk itself. I also tried allow=all in sip.conf  as well as
specifically listing allow= slin and all other formats to no avail. 

 

Also when this happens the channel is no longer usable even though Asterisk
thinks it is available. When the next call is placed it times out because
that channel has been locked by the above error. The only way out is a
complete reboot and reset of all systems. Not good. 

 

Any help would be greatly appreciated. If I had hair left I'd be pulling it
out about now.

 

 

Thanks

_____________________

 

Kevin Savoy

Business Unit Telecom Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

 <http://www.novo1.com/> http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 


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