Post your Asterisk's sip.conf<br><br><div class="gmail_quote">On Thu, Mar 11, 2010 at 10:39 PM, jonas kellens <span dir="ltr"><<a href="mailto:jonas.kellens@telenet.be">jonas.kellens@telenet.be</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div>
How can I set the prefered codec between 2 calling parties ??<br>
<br>
My Grandstream supports <font color="#0000ff"><b>G729, alaw and gsm</b></font>... in this order.<br>
The Zoiper softphone has <font color="#0000ff"><b>alaw and gsm</b></font> 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 color="#0000ff"><font size="2">[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 color="#0000ff"><font size="2">[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 color="#0000ff"><font size="2">[Mar 11 17:55:57] Peer audio RTP is at port <a href="http://192.168.1.101:10110" target="_blank">192.168.1.101:10110</a> </font></font><font color="#000000">(<-- the Grandstream)</font><br>
<font color="#0000ff"><font size="2">[Mar 11 17:55:57] Found audio description format PCMA for ID 8</font></font><br>
<font color="#0000ff"><font size="2">[Mar 11 17:55:57] Found audio description format GSM for ID 3</font></font><br>
<font color="#0000ff"><font size="2">[Mar 11 17:55:57] Found audio description format PCMU for ID 0</font></font><br>
<font color="#0000ff"><font size="2">[Mar 11 17:55:57] Found audio description format G729 for ID 18</font></font><br>
<font color="#0000ff"><font size="2">[Mar 11 17:55:57] Found audio description format telephone-event for ID 101</font></font><br>
<font color="#0000ff"><font size="2">[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 color="#0000ff"><font size="2">[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><font color="#888888">
<br>
Jonas.
</font></div>
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