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<DIV><FONT face=Arial size=2>Hi Guys, </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have following scenario which causes an issue
related to codecs (please look below)<BR><BR>[asterisk] -> Quintum CRSP* /
Quintum CMS -> PSTN <BR><BR>* Quintum Call Relay SP (CRSP - <A
href="http://www.quintum.com/main/servproducts.html?id=15">http://www.quintum.com/main/servproducts.html?id=15</A>),
Quintum CMS - H323 based gateway<BR><BR>When a call is being placed using a
SIP client, UAC (sip client) sends a list of supported
codecs to the asterisk and combined list is build as
supposed, <BR><BR>e.g.<BR></FONT></DIV>
<DIV><FONT face=Arial size=2><EM>Capabilities: us - 0x10d (g723|ulaw|alaw|g729),
peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c
(ulaw|alaw|g729)</EM><BR></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>My Asterisk has following codec settings
applied:</FONT></DIV>
<DIV> </DIV>
<DIV><EM>disallow=all<BR>allow=g723<BR>allow=g729<BR>allow=alaw<BR>allow=ulaw</EM></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Asterisk forwarding calls with prefix 00 to
Quintum SIP-PSTN (Quintum CRSP) gateway which has enabled g723,g729, ulaw
and alaw codecs - the same as ones on the asterisk side. <BR></FONT></DIV>
<DIV><FONT face=Arial size=2>Quintum CRSP sends back only the first codec out of
the supported codecs list on my side (asterisk) and within the combined
list there is <EM>only</EM> the first codec on asterisk side. Please
find below log out of described above behaviour:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><EM>Sip read:<BR>SIP/2.0 200 OK<BR>CSeq: 102
INVITE<BR>Call-ID: </EM><A
href=""><EM>4334482065fd376d226204fe4cdd3722@5.6.7.8</EM></A><BR><EM>Contact:
<sip:35923456@1.2.3.4><BR>Content-Type: application/sdp<BR>From:
"pavel"<sip:765@5.6.7.8:5060>;tag=as74cec9db<BR>To:
<sip:35923456@1.2.3.4>;tag=3ef4af85-1112b<BR>Via: SIP/2.0/UDP
5.6.7.8:5060;branch=z9hG4bK5568aa34<BR>Content-Length: 168<BR>User-Agent:
Quintum/1.0.0</EM></FONT></DIV>
<DIV><FONT face=Arial size=2><EM></EM></FONT> </DIV>
<DIV><FONT face=Arial size=2><EM>v=0<BR>o=Quintum 33034 31527 IN IP4
1.2.3.4<BR>s=VoipCall<BR>c=IN IP4 1.2.3.4<BR>t=0 0<BR>m=audio 10858 RTP/AVP
4<BR>c=IN IP4 1.2.3.4<BR>a=rtpmap:4 g723/8000/1</EM></FONT></DIV>
<DIV><FONT face=Arial size=2><EM></EM></FONT> </DIV>
<DIV><FONT face=Arial size=2><EM>10 headers, 8 lines<BR>Found RTP audio format
4<BR>Peer audio RTP is at port 1.2.3.4 :10858<BR>Found description format
g723<BR><FONT color=#ff0000>Capabilities: us - 0x10d
(<STRONG>g723</STRONG>|ulaw|alaw|g729), peer - audio=0x1 (g723)/video=0x0
(nothing), combined - 0x1 <STRONG>(g723)</STRONG></FONT></EM></FONT></DIV>
<DIV><EM><FONT face=Arial color=#ff0000 size=2></FONT></EM> </DIV>
<DIV><FONT face=Arial>where <EM>us</EM> is asterisk, <EM>peer</EM> is the
Quintum CRSP</FONT></DIV>
<DIV><EM><FONT face=Arial color=#ff0000 size=2></FONT></EM> </DIV>
<DIV><FONT face=Arial size=2>This way if the SIP client supports only G729 the
call fails since the combined list of codecs would be again g723 as long as the
g729 is not the first in the list of asterisk's codecs.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV>Is there any way I can forcably determine the codecs reported for the peer
out of the asterisk's codec capabilities list?<BR></DIV>
<DIV><FONT face=Arial size=2><EM>Any idea how I could force the asterisk to
change the order of supported codecs according to the SIP client first codec in
the list or how I could force the Quintum to send back the full list of
supported codecs ?</EM></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks and regards,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>Pavel</FONT></DIV></DIV><BR></FONT></DIV></FONT></DIV></BODY></HTML>