either<br>1)pay digium for g.729 license or<br>2)allow g.729 for sip3<br><br>- sip 1 -> sip2 work cause it will pass thru, <br>- sip 2 -> sip3 fails because since asterisk wants to do transcoding to 729<->711 and no license
<br>if bandwidth is a concern just use GSM (if available as a codec on the phone)<br><br><div><span class="gmail_quote">On 8/8/06, <b class="gmail_sendername">Chan Kwang Mien</b> <<a href="mailto:kwangmien@asgent-tech.com">
kwangmien@asgent-tech.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi,<br><br>My test-setup is as follows :<br><br>
sip1 <--> Asterisk <--> sip2<br> ^<br> |-------> sip3<br><br>In sip.conf,<br><br>[sip1]<br>type=friend<br>host=dynamic<br>secret=pass<br>disallow=all<br>allow=g729<br>allow=ulaw<br>
<br>[sip2]<br>type=friend<br>host=dynamic<br>secret=pass<br>disallow=all<br>allow=g729<br><br>[sip3]<br>type=friend<br>host=dynamic<br>secret=pass<br>disallow=all<br>allow=ulaw<br><br><br>sip1 supports g.729 and g.711u only
<br>sip2 supports g.729 only<br>sip3 supports g.711u only<br><br>sip1 is able to establish a call to sip2.<br>However, I have problem establishing a call from sip1 to sip3. sip3<br>rings but when I answered it, it hanged up.
<br><br>The Logs are :<br><br> -- Executing Dial("SIP/2006-389a", "SIP/2003") in new stack<br> -- Called 2003<br>Aug 8 09:55:15 WARNING[6937]: channel.c:2725<br>ast_channel_make_compatible: No path to translate from SIP/2003-b5f8(4)
<br>to SIP/2006-389a(256)<br><br> -- SIP/2003-b5f8 is ringing<br> -- SIP/2003-b5f8 answered SIP/2006-389a<br><br>Aug 8 09:55:16 WARNING[6937]: channel.c:2725<br>ast_channel_make_compatible: No path to translate from
<br>SIP/2006-389a(256) to SIP/2003-b5f8(4)<br>Aug 8 09:55:16 WARNING[6937]: app_dial.c:1608 dial_exec_full: Had to<br>drop call because I couldn't make SIP/2006-389a compatible with<br>SIP/2003-b5f8<br> == Spawn extension (phones, 2003, 1) exited non-zero on
<br>'SIP/2006-389a'<br><br><br>I think the codecs used by sip3 and sip1 are incompatible. Does anyone<br>know how I could make them compatible ?<br><br><br>Thank you.<br><br>Regards,<br>Kwang Mien<br><br><br><br>_______________________________________________
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