<span id="result_box" class="long_text"><span title="">Good afternoon
list.<br><br></span><span style="" title="">I'm experiencing a problem
with my SIP channel's. </span><span title="">When I have an external
connection for one of my SIP carrier's, I can listen to the
client and the client listens to me normally. </span><span style="background-color: rgb(255, 255, 255);" title="">The problem is
when I will transfer this connection, the call is mute for the extension I have transfered. </span><span title="">Only the client hears
normally. </span><span title="">In the console of Asterisk generates the
following warning:<br><br></span><span style="background-color: rgb(255, 255, 255);" title="">[Jul 19 14:46:24] WARNING [9220]:
chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while
native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) </span><span title="">/ 0x2 (gsm) (2)<br></span><span style="background-color: rgb(255, 255, 255);" title="">[Jul 19 14:46:24] WARNING [9220]:
chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while
native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) </span><span title="">/ 0x2 (gsm) (2)<br><br><br></span><span style="" title="">Detail,
this happens with both the codec gsm, ulaw, alaw and g729 and with any
of my SIP carrier's (I own three). </span><span style="background-color: rgb(255, 255, 255);" title="">And only happens
when the call is transferred.<br><br></span><span title="">Does anyone
have any idea what could be?<br><br>Thanks,<br>Rodrigo Lang.<br></span></span>