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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:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail
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.
On 6/28/06, Kevin Savoy <[EMAIL PROTECTED]> 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 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 Novo 1 is a service mark of Novo 1, Inc
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