Thanks for the response.<br>I have disallow=all and allow=alaw in sip.conf for my SIP user.<br>Any other idea?<br>--AM<br><br><div class="gmail_quote">On Tue, Jun 28, 2011 at 4:23 PM, Fellipe Paes <span dir="ltr"><<a href="mailto:fellipe_ps@hotmail.com">fellipe_ps@hotmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><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>Date: Tue, 28 Jun 2011 15:29:11 +0530<br>
From: <a href="mailto:theasteriskman@gmail.com" target="_blank">theasteriskman@gmail.com</a><br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>Subject: [asterisk-users] Asked to transmit frame type slin,        while native formats is 0x8 (alaw)<div>
<div class="h5"><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></div></div>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --
New to Asterisk? Join us for a live introductory webinar every Thurs:
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a>
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></div></div>                                            </div></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br>