Hello Steve (Fax god)!<div>Thanks for your explanation.</div><div>4 days ago I installed FS (git version) on my test server. Yesterday I've received my new SS7 connection. so I was testing & able to receive faxes (SIP-alaw to SIP-T.38) and I couldn't test send faxes because my Telco didn't activate outbound calling. my test was success at 30 concurrent calls/faxes.</div>
<div><br></div><div>Today my telco activate outbound calling and I've updated FS to today's git version. I did update (git pull && make current) without keeping a backup of old FS sources. While I was updating I had a same problem like "Update fail on mod_spandsp" finally I've updated the system. the problem is now I can't receive or send a fax with today's git version.</div>
<div><br></div><div>My sip log indicates I was using "User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28eb6e0 2010-07-21 00-22-24 +0800" and this version support receive faxes.</div><div>Now I'm using "User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-e704f02 2010-07-25 10-10-06 +0200" and this version doesn't allow me to send and receive faxes.</div>
<div><br></div><div>Please let me know how can I download "2010-07-21" version again. then I can continue my testing.</div><div><br></div><div>Thanks a lot.</div><div><br></div><div>Weera Suriya</div><div><br></div>
<div><br><div class="gmail_quote">On Tue, Jul 20, 2010 at 11:45 PM, Steve Underwood <span dir="ltr"><<a href="mailto:steveu@coppice.org">steveu@coppice.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><div></div><div class="h5">On 07/20/2010 11:16 PM, Weera Suriya wrote:<br>
> Hello list,<br>
> I have a question about t.38 fax gateway feature in FS(mod_spandsp).<br>
> here is my setup.<br>
> PSTN <----SS7--->YATE(doesn't support T.38 fax<br>
> gateway)<----IAX2--alaw---->CallWeaver-Faxgateway<------SIP--T.38------->OpenSIPs<------->ATAs(T.38)<---->fax<br>
> machines<br>
> I've 16 E1s connected with my Yate server and Yate server send faxes<br>
> to Callweaver servers (6 servers for CallWeaver) via IAX2(alaw) . then<br>
> CallWeaver servers send faxes to OpenSIPs server via SIP T.38 and<br>
> OpenSIPs server send SIP calls to ATAs<br>
><br>
> Now I want to expand this setup by adding 16 more E1 connections and I<br>
> want to replace these 6 CallWeaver servers with one or two FreeSwitch<br>
> :) I would like to know one thing. does FS able to convert SIP alaw<br>
> fax (or IAX2 alaw fax) into SIP T.38 fax (I know TDM to SIP t.38 works<br>
> with FS)<br>
><br>
</div></div>We have T.38 gateway functionality implemented in FS now. At this point<br>
it should be more robust than Callweaver in its call negotiation logic,<br>
and the logic in the T.38 engine. However, it has not yet been pushed as<br>
hard as Callweaver with concurrent channels. The design looks good, but<br>
we still need to prove no processing spikes, or similar issues, prevent<br>
the concurrent channels going right up to the limits of the processors.<br>
If you want to try high volumes with FS we will be grateful for any<br>
feedback, and I will consider addressing any performance issues you<br>
might find a priority.<br>
<font color="#888888"><br>
Steve<br>
</font><div><div></div><div class="h5"><br>
<br>
_______________________________________________<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
</div></div></blockquote></div><br></div>