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<DIV><SPAN class=401512713-08082006><FONT face=Arial color=#0000ff size=2>I have
the same problem here, why does asterisk not use ulaw with Sip1 -> Sip3
? As it has allow=g729 and allow=ulaw in Sip1, should it not fallback onto
ulaw when the g729 fails?</FONT></SPAN></DIV>
<DIV><SPAN class=401512713-08082006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=401512713-08082006><FONT face=Arial color=#0000ff
size=2>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=401512713-08082006><FONT face=Arial color=#0000ff
size=2>Dean.</FONT></SPAN></DIV>
<BLOCKQUOTE>
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]<B>On Behalf Of </B>Rosli
Sukri<BR><B>Sent:</B> 08 August 2006 13:38<BR><B>To:</B> Asterisk Users
Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[asterisk-users] Problems with Codecs in
Asterisk<BR><BR></FONT></DIV>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="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">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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