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<p class="MsoNormal">I don’t believe that the code that adds Required: timers to a 200 OK response will work, even in Asterisk 13, current branch version.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">In my back port, it produces an error saying headers cannot be added after lines have been added. The same conditions for this seem to apply in version 13:<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">In add_header:<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> if (req->lines) {<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> ast_log(LOG_WARNING, "Can't add more headers when lines have been added\n");<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> return -1;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> }<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""><o:p> </o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">In transmit_response_with_sdp:<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> if (p->rtp) {<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New"">...<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> if (p->t38.state == T38_ENABLED) {<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> add_sdp(&resp, p, oldsdp, TRUE, TRUE);<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> } else {<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> add_sdp(&resp, p, oldsdp, TRUE, FALSE);<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> }<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> } else<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> if (reliable && !p->pendinginvite)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> p->pendinginvite = seqno; /* Buggy clients sends ACK on RINGING too */<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Times New Roman",serif"> add_required_respheader(&resp);<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Times New Roman",serif"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Times New Roman",serif"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Times New Roman",serif">Note that SDP is added, and therefore “req->lines” becomes non-zero, as the response now has a body, before add_required_respheader is called.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Times New Roman",serif"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Times New Roman",serif">In add_required_respheader:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Times New Roman",serif"><o:p> </o:p></span></p>
<pre> if (ast_str_strlen(str) > 0) {<o:p></o:p></pre>
<pre> add_header(req, "Require", ast_str_buffer(str));<o:p></o:p></pre>
<pre> }<o:p></o:p></pre>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">For information, the test setup for this was two Asterisk instances in tandem. The first one uses session-timers=originate. The second one uses sendrpid=true, session-timers=accept and session-refresher=uas. The dialplan (after a delay)
sets the connected line, to force a re-invite against the call setup direction (then goes into a long wait). As the first Asterisk is UAS for the re-invite, but is in session timer originate mode, it is forced to use Require. The test was trying to confirm
that session timer role reversals worked.<o:p></o:p></p>
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