<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><span style="background-color: rgb(255, 255, 255); ">What Ken said, the prime is mismatched. </span></div><span style="background-color: rgb(255, 255, 255); "><div><span style="background-color: rgb(255, 255, 255); "><br></span></div>2013-04-18 21:27:24.357839 [DEBUG] sofia_glue.c:5155 Audio Codec Compare [G729:18:8000:<b><u>30</u></b>:8000]/[G729:18:8000:<b><u>20</u></b>:8000]</span><br><div><div><br></div><div>You should be able to change it on the client side. If you are having trouble with that, what endpoint are you using - Cisco SPAs, Grandstream, Polycom, etc?</div><div><br></div><div>Nick</div><div><br></div><div><br></div><div>On Apr 18, 2013, at 6:49 PM, Jimmy Chang <<a href="mailto:chang33.tw@gmail.com">chang33.tw@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">
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<div text="#000000" bgcolor="#FFFFFF">
<div class="moz-cite-prefix">Hi,<br>
<br>
Here is the FS 1.3.17 log.<br>
<br>
2013-04-18 21:27:24.357839 [DEBUG] sofia.c:5597 Channel
sofia/ivvr/9999110 entering state [proceeding][180]<br>
2013-04-18 21:27:24.357839 [DEBUG] sofia.c:5608 Remote SDP:<br>
v=0<br>
o=Dialogic_SIP_CCLIB 62148360 62148361 IN IP4 10.0.110.41<br>
s=Dialogic_SIP_CCLIB<br>
c=IN IP4 10.0.110.41<br>
t=0 0<br>
m=audio 49152 RTP/AVP 18 101<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=ptime:30<br>
<br>
2013-04-18 21:27:24.357839 [DEBUG] sofia_glue.c:5155 Audio Codec
Compare [G729:18:8000:30:8000]/[G729:18:8000:20:8000]<br>
2013-04-18 21:27:24.357839 [DEBUG] sofia_glue.c:5212 Substituting
codec G729@30i@8000h<br>
2013-04-18 21:27:24.357839 [DEBUG] sofia_glue.c:3110 Set Codec
sofia/ivvr/9999110 G729/8000 30 ms 240 samples 8000 bits<br>
2013-04-18 21:27:24.357839 [DEBUG] switch_core_codec.c:111
sofia/ivvr/9999110 Original read codec set to G729:18<br>
<br>
<br>
The same clients and the same conf, it runs okay.<br>
<br>
Any further advice?<br>
Jimmy<br>
<br>
<br>
於 2013/4/18 下午 10:05, Ken Rice 提到:<br>
</div>
<blockquote cite="mid:CD9567DD.6F32C%25krice@freeswitch.org" type="cite">
<title>Re: [Freeswitch-users] G729 codec INCOMPATIBLE_DESTINATION</title>
<font face="Monaco, Courier New"><span style="font-size:11pt">Ptime
mismatch forcing a transcode... <br>
<br>
<br>
On 4/18/13 8:41 AM, "Jimmy Chang" <<a moz-do-not-send="true" href="x-msg://1856/chang33.tw@gmail.com">chang33.tw@gmail.com</a>>
wrote:<br>
<br>
</span></font>
<blockquote><font face="Monaco, Courier New"><span style="font-size:11pt"> </span></font><span style="font-size:11pt"><font face="Arial">Hi,<br>
<br>
I used FreeSWITCH Version
1.3.17+git~20130312T033518Z~7d29a92f55 and client could talk
with G729 without problem.<br>
Today I install a new FS and the version is <br>
FreeSWITCH Version 1.5.1b+git~20130417T010754Z~538d9beacd.<br>
<br>
I copied the entire conf directory in old FS to new FS.<br>
I got the problem.<br>
<br>
2013-04-18 21:25:06.899049 [DEBUG] sofia.c:5694 Remote
SDP:
<br>
v=0
<br>
o=Dialogic_SIP_CCLIB 62147888 62147889 IN IP4
10.0.110.41
<br>
s=Dialogic_SIP_CCLIB
<br>
c=IN IP4
10.0.110.41
<br>
t=0
0
<br>
m=audio 49152 RTP/AVP 18
101
<br>
a=fmtp:18
annexb=no
<br>
a=rtpmap:101
telephone-event/8000
<br>
a=fmtp:101
0-15
<br>
a=ptime:30
<br>
<br>
2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2597
Audio Codec Compare
[G729:18:8000:30:8000]/[G729:18:8000:20:8000] <br>
2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2597
Audio Codec Compare
[telephone-event:101:8000:30:0]/[G729:18:8000:20:8000]<br>
2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2671
Substituting codec
G729@30i@8000h <br>
2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:1617
No audio codec
available <br>
2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2752
Set 2833 dtmf send payload to
101 <br>
2013-04-18 21:25:06.899049 [NOTICE] sofia.c:5812 Hangup
sofia/ivvr/9999110 [CS_CONSUME_MEDIA]
[INCOMPATIBLE_DESTINATION] <br>
<br>
<br>
Any advice?<br>
<br>
Thanks.<br>
Jimmy<br>
<br>
</font><font face="Monaco, Courier New"> <br>
<hr size="3" width="95%" align="CENTER"></font></span><font size="2"><font face="Consolas, Courier New, Courier"><span style="font-size:10pt">_________________________________________________________________________<br>
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<font size="2"><font face="Consolas, Courier New, Courier"><span style="font-size:10pt"><br>
</span></font></font><font face="Monaco, Courier New"><span style="font-size:11pt">-- <br>
Ken<br>
<font color="#0000FF"><u><a moz-do-not-send="true" href="http://www.freeswitch.org/">http://www.FreeSWITCH.org</a><br>
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</u></font><a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br>
</span></font>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_________________________________________________________________________
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_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>http://www.freeswitchsolutions.com<br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>http://www.cudatel.com<br><br>Official FreeSWITCH Sites<br>http://www.freeswitch.org<br>http://wiki.freeswitch.org<br>http://www.cluecon.com<br><br>FreeSWITCH-users mailing list<br>FreeSWITCH-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></body></html>