Good afternoon list. I'm experiencing a problem with my SIP channel's. 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. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the following warning:
[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) / 0x2 (gsm) (2) [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) / 0x2 (gsm) (2) Detail, this happens with both the codec gsm, ulaw, alaw and g729 and with any of my SIP carrier's (I own three). And only happens when the call is transferred. Does anyone have any idea what could be? Thanks, Rodrigo Lang.
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