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Sip.conf :<BR>
<BR>
[general]<BR>
;context=default<BR>
allowguest=no<BR>
allowoverlap=no<BR>
allowtransfer=yes<BR>
realm=mydomain<BR>
bindport=5060<BR>
bindaddr=X.X.X.X<BR>
maxexpiry=1800 <BR>
minexpiry=60 <BR>
mohinterpret=default<BR>
mohsuggest=default<BR>
language=be<BR>
useragent=mycorp<BR>
dtmfmode = rfc2833 <BR>
alwaysauthreject = yes <BR>
;contactdeny=0.0.0.0/0.0.0.0<BR>
;contactpermit=172.16.0.0/255.255.0.0<BR>
rtptimeout=60<BR>
rtpholdtimeout=300<BR>
;sipdebug = yes<BR>
;recordhistory=yes<BR>
;dumphistory=yes <BR>
registertimeout=60<BR>
registerattempts=60 <BR>
rtcachefriends=yes <BR>
;rtsavesysname=yes <BR>
;rtupdate=yes <BR>
;rtautoclear=yes <BR>
;ignoreregexpire=yes<BR>
jbenable = yes<BR>
jbforce = no<BR>
allowsubscribe=yes<BR>
limitonpeer = yes<BR>
notifyringing=yes<BR>
notifyhold=yes<BR>
<BR>
then come the registrations...<BR>
<BR>
<BR>
Jonas.<BR>
<BR>
<BR>
On Fri, 2010-03-12 at 11:47 +0530, Prince Singh wrote:<BR>
<BLOCKQUOTE TYPE=CITE>
Post your Asterisk's sip.conf<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
On Thu, Mar 11, 2010 at 10:39 PM, jonas kellens <<A HREF="mailto:jonas.kellens@telenet.be">jonas.kellens@telenet.be</A>> wrote:
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BLOCKQUOTE>
How can I set the prefered codec between 2 calling parties ??<BR>
<BR>
My Grandstream supports <B><FONT COLOR="#0000ff">G729, alaw and gsm</FONT></B>... in this order.<BR>
The Zoiper softphone has <B><FONT COLOR="#0000ff">alaw and gsm</FONT></B> as codecs... in that order.<BR>
<BR>
Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone.<BR>
<BR>
CLI shows :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to translate from SIP/mygrandstream-09c599e0(2) to SIP/zoiper-09cd57f8(256)</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:47:21] -- Got SIP response 415 "Unsupported Media Type" back from 192.168.1.106</FONT></FONT> (<-- zoiper)<BR>
<BR>
SIP debug :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Peer audio RTP is at port <A HREF="http://192.168.1.101:10110">192.168.1.101:10110</A> </FONT></FONT><FONT COLOR="#000000">(<-- the Grandstream)</FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Found audio description format PCMA for ID 8</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Found audio description format GSM for ID 3</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Found audio description format PCMU for ID 0</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Found audio description format G729 for ID 18</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Found audio description format telephone-event for ID 101</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10a (gsm|alaw|g729)</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)</FONT></FONT><BR>
...<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Audio is at 192.168.1.150 port 11586 </FONT></FONT><FONT COLOR="#000000">(<-- my Asterisk)</FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Adding codec 0x100 (g729) to SDP</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 11 17:55:57] Adding non-codec 0x1 (telephone-event) to S</FONT></FONT>DP<BR>
<BR>
This is what Asterisk sends to the Zoiper in the INVITE (sdp) :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">Content-Type: application/sdp</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">Content-Length: 263</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">v=0</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">o=root 3208 3208 IN IP4 192.168.1.150</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">s=session</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">c=IN IP4 192.168.1.150</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">t=0 0</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">m=audio 11586 RTP/AVP 18 101</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">a=rtpmap:18 G729/8000</FONT></FONT><BR>
<BR>
<FONT COLOR="#000000">Why isn't Asterisk negotiating with the Zoiper for the alaw-codec ??</FONT><BR>
<BR>
<FONT COLOR="#000000">The sip-configuration (realtime</FONT> MySQL<FONT COLOR="#000000">) for the Grandstream is :</FONT><BR>
<BR>
<FONT COLOR="#0000ff">allow : </FONT><FONT COLOR="#0000ff"><B>g729;alaw;gsm</B></FONT><BR>
<BR>
and the Zoiper softphone :<BR>
<BR>
<FONT COLOR="#0000ff">allow : </FONT><FONT COLOR="#0000ff"><B>alaw;gsm;g729</B></FONT><BR>
<BR>
<BR>
Kind regards,<BR>
<BR>
<FONT COLOR="#888888">Jonas.</FONT>
</BLOCKQUOTE>
</BLOCKQUOTE>
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