[Asterisk-Users] ATA 186 AudioMode setting for a call to PSTN with X100P

[Dan Fernandez]

I have a problem with the AudioMode configuration on an ATA 186 V2.16 and *

I have set the ATA to use preferably g723.1 (LBRCodec=0, RxCodec=0, TxCodec=0)

Bit 1 of AudioMode concerns with 

0/1  Enable/Disable low-bit-rate codec. In addition to using the G.711 codec, the Cisco ATA can use a low-bit-rate codec

If I have AudioMode= 0x00140014 (that is with Bit 1 enabled, I can place a call between my ATA and MSN with g723.1) I can also place calls to voicemail or other * apps (with a previous change to g711 via SIP_CODEC=ulaw.).   However, if I try to place a call to the PSTN via my X100P  it doesn´t work:

Based on the sip dump, with the Audiocode set to 0x00140014, * doesn´t do a codec change to ulaw as I requested via de SIP_CODEC. 

It works fine for a call originated from MSN. From this respect it appears the way to fix it would be to change the values of AudioMode. However, if I change AudioMode to 0x00120012 I get the opposite. I can place a call to the PSTN via a Zap channel but I cannot call other SIP client via g723.. 

What can I do to be able to get both things to work?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030508/593b0a9a/attachment.htm