<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style>
</head>
<body class='hmmessage'><div dir='ltr'>
Hello!<div><br></div><div>In your sip.conf use alaw as your first codec option and see what happens.</div><div>Best regards,</div><div><br></div><div>Fellipe Paes<br><br><div><hr id="stopSpelling">Date: Tue, 28 Jun 2011 15:29:11 +0530<br>From: theasteriskman@gmail.com<br>To: asterisk-users@lists.digium.com<br>Subject: [asterisk-users] Asked to transmit frame type slin,        while native formats is 0x8 (alaw)<br><br><br>Asterisk 1.8.3.2<br><br>I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. <br>I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string.<br>
If I call PSTN from SIP directly or vice versa I don't see this warning coming.<br>On SIP I have allowed only one codec(alaw).<br><br>[Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw)<br>
<br><br>I also tried to yes/no option transcode_via_sln in asterisk.conf without any success.<br>Any idea?<br>Thanks,<br>--AM<br>
<br>--
_____________________________________________________________________
-- 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</div></div>                                            </div></body>
</html>