correct, we *do not* proxy re-invites except when bypass_media or proxy_media is set.<br><br><br><div class="gmail_quote">On Tue, Mar 31, 2009 at 11:59 AM, Gabriel Kuri <span dir="ltr"><<a href="mailto:gkuri@ieee.org">gkuri@ieee.org</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;">> This part is interesting, and the subject of a discussion we had<br>
> recently. A number of systems try that second re-invite after a 488, but<br>
> the SIP specs say the call is pretty much dead after the 488 message is<br>
> exchanged. Are they just hoping that maybe the other end will be<br>
> non-compliant enough to keep the call alive, and recover its media mode,<br>
> or haven't they read the specs?<br>
<br>
I think they're hoping the other end is willing to recover it's media<br>
mode rather than fail the call. I had no idea the call is technically<br>
dead after the 488. Honestly, it would be nice if the call would still<br>
be recoverable after that 488 on the T.38 ReINVITE, in order to try and<br>
negotiate PCMU to try and keep the FAX going, but if that's not how it's<br>
supposed to work, I'd rather follow the spec.<br>
<br>
So for now I've disabled T.38 completely on both the SPA side and I had<br>
the carrier disable it on my trunk, so they won't try a T.38 reinvite.<br>
Instead they're trying a PCMU ReINVITE and the problem I'm seeing is<br>
that if the carrier reINVITEs PCMU after the call initially started out<br>
as G729, FS fails the call, because it seems to be trying to transcode<br>
between G729 and PCMU, rather than pass the PCMU reINVITE through to the<br>
other leg.<br>
<br>
2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1550<br>
sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU<br>
2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1601<br>
sofia_glue_tech_set_codec() Set Codec<br>
sofia/<a href="http://cedarwireless.net/1909XXXXXXX@65.98.2" target="_blank">cedarwireless.net/1909XXXXXXX@65.98.2</a><br>
36.38 PCMU/8000 20 ms 160 samples<br>
2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp()<br>
Audio params are unchanged for sofia/<a href="http://cedarwireless.net/" target="_blank">cedarwireless.net/</a><br>
<a href="mailto:1909XXXXXXX@65.98.236.38">1909XXXXXXX@65.98.236.38</a>.<br>
2009-03-31 00:30:50 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state()<br>
Processing Reinvite<br>
2009-03-31 00:30:50 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state()<br>
Channel sofia/<a href="http://cedarwireless.net/1909XXXXXXX@65.98.236.38" target="_blank">cedarwireless.net/1909XXXXXXX@65.98.236.38</a> en<br>
tering state [completed]<br>
2009-03-31 00:30:50 [DEBUG] switch_core_io.c:655<br>
switch_core_session_write_frame()<br>
sofia/<a href="http://cedarwireless.net/1909XXXXXXX@65.98.23" target="_blank">cedarwireless.net/1909XXXXXXX@65.98.23</a><br>
6.38 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]<br>
2009-03-31 00:30:50 [ERR] mod_g729.c:145 switch_g729_decode() This codec<br>
is only usable in passthrough mode!<br>
2009-03-31 00:30:50 [ERR] switch_core_io.c:723<br>
switch_core_session_write_frame() Codec G.729 decoder error!<br>
<br>
I have inherit_codec=true, set in the dialplan and<br>
disable-transcoding=true set in the sofia profile, which is what I<br>
thought would do the trick, but it doesn't seem to be doing anything, FS<br>
is still trying to transcode between G729 and PCMU. Is there something<br>
I'm missing to get the PCMU ReINVITE from one of the legs to passthrough<br>
to the other leg? Does this work only in proxy_media or bypass_media modes?<br>
<br>
I am testing this with the latest rev of trunk as well.<br>
<br>
Thanks,<br>
<br>
Gabe<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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