[asterisk-dev] sip_write

Tzafrir Cohen tzafrir.cohen at xorcom.com
Tue Oct 13 02:34:01 CDT 2009

On Tue, Oct 13, 2009 at 11:39:32AM +0530, Chandrakant Solanki wrote:
> Hi
> While I am mute/unmute user from Manager...
> On asterisk CLI i found below error while playing ".gsm" file
> sip_write: Asked to transmit frame type 64, while native formats is 0x2
> (gsm)(2) read/write = 0x40 (slin)(64)/0x2 (gsm)(2)

This is a -users question. Please follow-up on asterisk-users (see
reply-to address.

You don't have the module codec_gsm.so ?

What is the output of 'core show translations' ?

               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir

More information about the asterisk-dev mailing list