<br><br><div><span class="gmail_quote">On 8/8/06, <b class="gmail_sendername">Dean @ INKnBITs</b> <<a href="mailto:dean.bath@inknbits.co.uk">dean.bath@inknbits.co.uk</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;">
<div>
<div>
<div><span><font color="#0000ff" face="Arial" 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></div></blockquote><div><br>true, it might be a problem on da sip phones itself (order of codec preference/precedence maybe) - can you confirm what codec is sip1 passing it to asterisk?..
<br><br>currently for me i am using a pa1688 based sip phone and when setting the codec you have to set the precedence order. i.e try ulaw, gsm then as a last option use 729.<br><br>i am speculating in this particular scenario during the initialisation of sip1 <-> asterisk wants bof of them probably agreed to do 729 as a result of the precedence setting on the phone
<br><br>maybe as an experiment, get sip3 to call sip2?<br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div><div><span><font color="#0000ff" face="Arial" size="2">
Thanks,</font></span></div></div><div><span class="sg">
<div><span><font color="#0000ff" face="Arial" size="2">Dean.</font></span></div></span></div><div><span class="e" id="q_10cedfe2308f0a6e_2">
<blockquote>
<div dir="ltr" align="left"><font face="Tahoma" size="2">-----Original Message-----<br><b>From:</b>
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>]<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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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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