<br><div class="gmail_quote">On Thu, Nov 5, 2009 at 8:23 AM, Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im"><br>
</div>We need to see how you are originating the calls; it's up to the<br>
originator to specify the formats that will be allowed for that call. In<br>
spool files, for example, there is a header that can be included to<br>
specify which audio (and video) codecs should be offered on the outgoing<br>
channel.<br><br></blockquote><div><br>Thanks Kevin, I was unaware of the Codecs header for the spool file.<br><br>However Asterisk still appears to be less than satisfied when asked to initiate a call with 'siren14' as the *only* "codec". (Obviously it isn't yet a full codec for Asterisk and is only a supported format. I suspect that is the key to this observation)<br>
<br>As a clean test, I did the following on a fresh install of CentOS:<br><br><span style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Arial; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px;"><div>
<span>svn checkout <a href="http://svn.digium.com/svn/asterisk/trunk">http://svn.digium.com/svn/asterisk/trunk</a> asterisk</span></div><div>cd asterisk<br>./configure</div><div>make menuselect <br></div><div>make install</div>
<div>make samples</div></span><br><span style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Arial; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px;">cp /usr/local/src/asterisk/contrib/init.d/rc.redhat.asterisk /etc/init.d/asterisk</span><br>
<br>asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv | grep siren<br> == Registered file format siren7, extension(s) siren7<br> format_siren7.so => (ITU G.722.1 (Siren7, licensed from Polycom))<br>
== Registered file format siren14, extension(s) siren14<br> format_siren14.so => (ITU G.722.1 Annex C (Siren14, licensed from Polycom))<br><br>(first make sure basic spool call works)<br><br>vi /etc/asterisk/sip.conf<br>
disallow=all<br> allow=ulaw<br><br>service asterisk restart<br><br>vi call.txt<br> Channel: SIP/<a href="mailto:foo@bar.com">foo@bar.com</a><br> CallerID: testcall<br> Context: default<br> Extension: demo<br>
Codecs: ulaw<br><br> cp call.txt /var/spool/asterisk/outgoing/<br><br>!!!! Outgoing INVITE sent to the folks at <a href="http://bar.com">bar.com</a> !!!!<br><br>(now let's try just siren14)<br><br>vi /etc/asterisk/sip.conf<br>
disallow=all<br>
allow=siren14<br>
<br>service asterisk restart<br><br>vi call.txt<br> Channel: SIP/<a href="mailto:foo@bar.com">foo@bar.com</a><br>
CallerID: testcall<br>
Context: default<br>
Extension: demo<br>
Codecs: siren14<br><br>cp call.txt /var/spool/asterisk/outgoing/<br><br> -- Attempting call on SIP/<a href="mailto:foo@bar.com">foo@bar.com</a> for demo@default:1 (Retry 1)<br>[Nov 10 07:43:13] WARNING[27630]: chan_sip.c:5735 sip_call: No audio format found to offer. Cancelling call to foo<br>
<br>So while inbound calls work fine with siren14 as the only allow=, Asterisk won't initiate an outbound call with siren14 as the only choice.<br><br>Tom<br><br><br><br> </div></div>