<br><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<font size="2">What are you reaching out to exactly? It would need to be a Siren14 capable. Also, do you have the Siren codec binary installed? It's not part of the Asterisk distribution.<br></font></blockquote><div>
<br>Inbound calls to Asterisk work (from a platform that supports both Siren14 and G.711). Leaving ulaw out of the allow list forces the inbound call to negotiate Siren14. In terms of debugging the outbound call, what I'm calling hasn't yet come into play as the error "No audio format found to offer. Cancelling call to blahblah" happens before any INVITE is transmitted from Asterisk.<br>
<br>Adding ulaw back into the allow list causes the outbound call to actually transmit the INVITE, but then they negotiate ulaw and not Siren14. Replacing ulaw with alaw in the allow list (a codec that is NOT supported by the platform I am calling) will also allow the outbound INVITE to be sent but no suitable codec is negotiated. Again, inbound calls do negotiate successfully.<br>
<br>The difference seems to be related to 'jointcapabilities' vs. 'capabilities' in the chan_sip.c <br><br>Are there other configuration settings I can adjust to negotiate Siren14 on outbound calls without hacking chan_sip.c?<br>
<br>I'm hoping I've completely missed something simple.<br><br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><font size="2">
Also, you should know that all Siren14 calls are presently downsampled to 16 KHz, so are effectively Siren7.Asterisk doesn't presently support sample rates beyond 16 KHz.<br></font></blockquote><div><br>format_siren14.c clearly seems to state support for the 48Kbps 32Khz Siren14 flavor and Kevin Flemming's earlier reply to this thread implied support for same.<br>
<br>Tom<br></div></div>