Rodrigo Lang schrieb: > 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. hello rodrigo,
this is exactly the problem i had. Have a look at issue 17641 (https://issues.asterisk.org/view.php?id=17641) There is a patch for asterisk 1.6.2.9 but its only a single row so you could easy find the position in app_dial.c to patch it by your own. the problem only occurs when you use answer in your dialplan. without an answer this wont happen. best regards. steve -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users