Hi, thanks a lot by the answers. But without the application Answer() the problem remains.
Realized over a battery of tests and refined the problem. Follows: A = External link that came with my Voip number. B = Operator. C = The extent to which A want to speak. A called my number and B answer. If B try to transfer with blindxfer (#) to C works fine. But if B try to transfer with atxfer (*2) he can talk to C, only when B hangs up to complete the transfer begins to generate those warnings on the cli. After the transfer using C atxfer not hear A, but A hears C. I believe it has become clearer now. And as he said, with any codec, and only when the person connects to my VoIP trunks. I did the test with the analogue trunks and atxfer worked normal. Thanks, Rodrigo Lang. 2010/7/20 Stefan Schmidt <[email protected]> > 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 >
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