<div>
<div>Hi Luigi,</div>
<div> </div>
<div>I am experiencing similar problem, i don't know if they are related. I even tried changing the line in chan_sip.c that you suggested, but it did not help.</div>
<div>I first noticed this when i was testing on Asterisk SVN-trunk-r15187 to use the PARKEDAT variable feature.</div>
<div> </div>
<div>
<div>- I park the call using ParkAndAnnounce</div>
<div>- plays moh.</div>
<div>- accept the call using ParkedCall</div>
<div> </div>
<div>The following errors are coming on the console and there is oneway audio - no audio after Music-On-Hold at caller's side. I am testing using cisco 7902 phones and using cisco 2800 router. Codec is g711ulaw</div>
<div> </div>
<div>Could you find any solution for the oneway audio ??</div>
<div> </div>
<div>thanks,</div>
<div>Sharath Chandra</div>
<div>================================================================ </div>
<div> </div>
<div>-- Executing ParkedCall("SIP/192.168.50.2-09cbd610", "366")<br> -- Channel SIP/192.168.50.2-09cbd610 connected to parked call 366<br>Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)
<br>Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)<br>Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)
<br>Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)<br>Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)
<br>Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)<br> </div></div>
<div> </div></div>
<div> </div>
<div> </div>
<div> </div>
<div><br><br> </div>
<div><span class="gmail_quote">On 3/29/06, <b class="gmail_sendername">Luigi Rizzo</b> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:rizzo@icir.org" target="_blank">rizzo@icir.org</a>> wrote:
</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">bug found: for the original version that introduced<br>the problem, the patch is below.<br><br>Index: channels/chan_sip.c
<br>===================================================================<br>--- channels/chan_sip.c (revision 15187)<br>+++ channels/chan_sip.c (working copy)<br>@@ -7244,7 +7244,7 @@<br> }<br> if (!(res = check_auth(p, req, user->name, user->secret, user->md5secret, sipmethod, uri, reliable, ignore))) {
<br> sip_cancel_destroy(p);<br>- ast_copy_flags(&p->flags[0], &user->flags[1], SIP_FLAGS_TO_COPY);<br>+ ast_copy_flags(&p->flags[0], &user->flags[0], SIP_FLAGS_TO_COPY);
<br> ast_copy_flags(&p->flags[1], &user->flags[1], SIP_PAGE2_FLAGS_TO_COPY);<br> /* Copy SIP extensions profile from INVITE */<br> if (p->sipoptions)
<br><br>On Tue, Mar 28, 2006 at 08:46:03AM -0800, Luigi Rizzo wrote:<br>> not sure if there is something local that is not handled<br>> properly, but in the transition from SVN 15186 -> 15187<br>> (a large change to chan_sip.c) i have lost the audio
<br>> at least on SIP calls, and apparently newer versions<br>> of chan_sip.c (up to 15610 which is HEAD now) don't fix the problem.<br>><br>> On the other hand, everything at SVN15610 and chan_sip.c.15186<br>
> gives me the audio so the bug must be there.<br>><br>> I am pretty sure it is something trivial;<br>> however, probably who made that change might have<br>> a more clear idea of what is going on ?<br>>
<br>> Are others seeing this too ?<br>><br>> thanks<br>> luigi<br>> _______________________________________________<br>> --Bandwidth and Colocation provided by <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://easynews.com/" target="_blank">
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