Hi, thanks a lot by the answers. But without the application Answer() the problem <span id="result_box" class="long_text"><span title="">remains.<br><br><br></span></span><span class="goog-zippy-collapsed" tabindex="0" id="romanspan" style=""><span id="romantext"></span></span><div id="gt-res-content" class="almost_half_cell" style="">
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<span id="result_box" class="long_text"><span title="">Realized over a battery of tests and
refined the problem. </span><span title="">Follows:<br><br></span><span title="">A = External link that came with my Voip number.<br></span><span title="">B = Operator.<br></span><span title="">C = The extent to which A want to speak.<br>
<br></span><span title="">A called my number
and B answer. </span><span title="">If B try to transfer with
blindxfer (#) to C works fine. </span><span style="background-color: rgb(255, 255, 255);" title="">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. </span><span style="background-color: rgb(255, 255, 255);" title="">After the
transfer using C atxfer not hear A, but A hears C.<br><br></span><span style="background-color: rgb(255, 255, 255);" title="">I believe it has
become clearer now. </span><span style="background-color: rgb(255, 255, 255);" title="">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.<br><br><br></span><span title="">Thanks,<br></span><span title="">Rodrigo Lang.</span></span></div></div><br><span id="result_box" class="long_text"><span title=""><br></span></span><br><div class="gmail_quote">
2010/7/20 Stefan Schmidt <span dir="ltr"><<a href="mailto:sst@sil.at">sst@sil.at</a>></span><br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Rodrigo Lang schrieb:<br>
<div><div></div><div class="h5">> Good afternoon list.<br>
><br>
> I'm experiencing a problem with my SIP channel's. When I have an<br>
> external connection for one of my SIP carrier's, I can listen to the<br>
> client and the client listens to me normally. The problem is when I<br>
> will transfer this connection, the call is mute for the extension I<br>
> have transfered. Only the client hears normally. In the console of<br>
> Asterisk generates the following warning:<br>
><br>
> [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to<br>
> transmit frame type 64, while native formats is 0x2 (gsm) (2) read /<br>
> write = 0x40 (slin) (64) / 0x2 (gsm) (2)<br>
> [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to<br>
> transmit frame type 64, while native formats is 0x2 (gsm) (2) read /<br>
> write = 0x40 (slin) (64) / 0x2 (gsm) (2)<br>
><br>
><br>
> Detail, this happens with both the codec gsm, ulaw, alaw and g729 and<br>
> with any of my SIP carrier's (I own three). And only happens when the<br>
> call is transferred.<br>
><br>
> Does anyone have any idea what could be?<br>
><br>
> Thanks,<br>
> Rodrigo Lang.<br>
</div></div>hello rodrigo,<br>
<br>
this is exactly the problem i had. Have a look at issue 17641<br>
(<a href="https://issues.asterisk.org/view.php?id=17641" target="_blank">https://issues.asterisk.org/view.php?id=17641</a>)<br>
There is a patch for asterisk 1.6.2.9 but its only a single row so you<br>
could easy find the position in app_dial.c to patch it by your own.<br>
the problem only occurs when you use answer in your dialplan. without an<br>
answer this wont happen.<br>
<br>
<br>
best regards.<br>
<br>
steve<br>
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