[asterisk-users] Problem with SIP

[Rodrigo Lang]

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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