I don't think there is any way to do this without an RTP capable device in the mix.<div><br></div><div>What you may be able to do is have asterisk detect that it's a fax, then reject it if it is.. I don't know if you can do all that without answering the call.</div>
<div><br></div><div>Then you can forward it back to the proxy if it is a fax with maybe a prefix. </div><div><br></div><div>A lot of assumptions in there. Would like to hear if you find something that works. Not sure if you can SIP Spiral yet in asterisk anyway. ;)</div>
<div>-Brett</div><div><br></div><div><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 10:51 AM, David J. <span dir="ltr"><<a href="mailto:david@styleflare.com">david@styleflare.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Matt,<br>
<br>
I am for sure probably wrong, but I think you would need Asterisk or<br>
Variant to Determine that it is a Fax Call,<br>
I dont think UAC's send T38 information without negotiating with the<br>
other side who request that it is capable, then it brings you to Jeff's<br>
answer.<br>
<br>
See above.<br>
<div class="im"><br>
<br>
Matthew S. Crocker wrote:<br>
> Can OpenSIPS make routing decisions based on the SDP information in an INVITE?<br>
><br>
> Lets say I have the following config<br>
><br>
> PSTN -> t.38 Gateway -> OpenSIPS -> UserAgent<br>
><br>
> I have a TN from the PSTN routed to the UserAgent, I'd like to provide a service so the user can use the TN for both voice & faxing.<br>
><br>
> Voice call goes through normally (g.711 g.729 codec)<br>
><br>
> Fax call starts off as a normal voice call (INVITE, 180, 183, 200). Once the call is answered the originating end (PSTN) starts sending fax tones. The Gateway hears the fax tones and attempts to RE-INVITE with T.38 in the SDP. I'd like OpenSIPS to see the T.38 capability in the SDP and redirect the call to a fax->e-mail gateway. So, the 2nd INVITE comes in, OpenSIPS sends the INVITE to the fax gateway and a BYE to the user. The fax gateway does a 200 and negotiates T.38 with the PSTN gateway.<br>
><br>
> I know I can route the call through Asterisk and have it do a quiet answer and listen for the modem sounds. I'd like to avoid using Asterisk for all RTP traffic and only use it for the fax gateway traffic (i.e. once it has been determined to be a fax Asterisk steps in and handled the T38 -> E-mail)<br>
><br>
> -Matt<br>
><br>
><br>
<br>
<br>
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