[asterisk-dev] RxFax, SpanDSP & Asterisk possible bug?

[Steve Underwood]

I suspect the problem is you can't read.

I hate people saying this over and over, but now I'm going to say it - 
this isn't a development question, its a user problem. Wrong mailing list.

Steve

Sebastian wrote:

> Hi,
>
> For some time now, I've been fighting with RxFax and Asterisk.
> I had it working for some time, however, for some reason it just 
> stopped working, I guess someone updated Asterisk or something, don't 
> know exactly.
>
> At the moment I keep getting errors while entering the RxFax stage of 
> a call.
> But due to the fact RxFax does not contain any code to directly 
> interact with an RTP stream, it uses SpanDSP, I guess the problem is 
> somewhere in SpanDSP.
>
> From my logs, I get:
>   -- Goto (macro-faxreceive,s,7)
>   -- Executing RxFAX("SIP/<faxnumber removed>-ff83", 
> "/var/spool/asterisk/fax/fax/asterisk-8558-1149842081.2.tif") in new 
> stack
> Jun  9 10:34:56 WARNING[8558]: chan_sip.c:1829 sip_write: Asked to 
> transmit frame type 64, while native formats is 4 (read/write = 64/4)
> Jun  9 10:34:56 WARNING[8558]: chan_sip.c:1829 sip_write: Asked to 
> transmit frame type 64, while native formats is 4 (read/write = 64/4)
> Jun  9 10:34:56 WARNING[8558]: chan_sip.c:1829 sip_write: Asked to 
> transmit frame type 64, while native formats is 4 (read/write = 64/4)
>
> The basic call setup works perfectly, but whenever I try to enter a 
> fax stage, I get this error.
> According to the SIP call initialisation, there are no problems with 
> formats, and the agreed upon format is 4 (aka ulaw).
> At the moment I get these errors, there aren't any SIP transmissions, 
> so were in mid-session...
>
> I guess the problem is somewhere in SpanDSP or a transcoding error 
> between slin and ulaw in chan_sip.
>
> I've confirmed this problem on 2 servers so far.
> The first server is our fax receiver (production), where this problem 
> started. But it also appeared when I tried to directly connect a fax 
> transmission from a PSTN line with an Sipura 3000 as FXO (forwarding 
> the audio stream over our NAT to our local Asterisk server, with the 
> ulaw codec).
> Both servers are Debian Sarge servers.
>
> Confirmed this problem in:
> Asterisk 1.0.7 (default .deb packages, being with SpanDSP 0.0.2pre10-3)
> Asterisk 1.0.11 (after reading 
> http://lists.digium.com/pipermail/asterisk-users/2006-May/151843.html, 
> SpanDSP 0.0.2pre26)
> Asterisk 1.2.8 (from source, with SpanDSP 0.0.2pre26)
> Asterisk 1.2.7.1 (from source, with SpanDSP 0.0.2pre25)
>
> Does anyone know of a way to get it working again?
>
> It has worked before, so I know it's perfectly possible with Debian 
> Sarge.
> Furthermore I've set this up on Red Hat Enterprise servers without 
> this problem (with Asterisk 1.0.9). So I don't believe this is an 
> error caused by myself.
>