[asterisk-users] Asterisk 1.6.1.13 and T.38 faxing

Vinícius Fontes vinicius at canall.com.br
Tue Feb 2 13:18:52 CST 2010


----- "Steve Underwood" <steveu at coppice.org> escreveu:

> On 02/03/2010 12:45 AM, Vinícius Fontes wrote:
> > ----- "Kevin P. Fleming"<kpfleming at digium.com>  escreveu:
> >
> >    
> >> Vinícius Fontes wrote:
> >>
> >>      
> >>> I couldn't agree more Steve.
> >>>
> >>> Is there any other info I could provide in order to help you find
> >>>        
> >> out what's wrong? I could even open an issue on Mantis if the
> Digium
> >> staff think it's worth it.
> >>
> >> Post a 'sip set debug' capture of the failing call in this thread;
> >> that
> >> will make it much more obvious what is happening.
> >>
> >> -- 
> >> Kevin P. Fleming
> >> Digium, Inc. | Director of Software Technologies
> >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> >> skype: kpfleming | jabber: kpfleming at digium.com
> >> Check us out at www.digium.com&  www.asterisk.org
> >>
> >> -- 
> >>
> _____________________________________________________________________
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> >>
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> >>      
> >
> > I've put it on pastebin because is was a lot of text. Here's the
> link: http://pastebin.com/m7467cea1. That's all the information on the
> CLI with verbose=3 and "sip set debug peer voxip".
> >
> >    
> I wonder why Asterisk would say:
> 
> X-asterisk-Info: SIP re-invite (External RTP bridge)
> Content-Type: application/sdp
> Content-Length: 344
> 
> v=0
> o=root 44350963 44350964 IN IP4 10.153.66.146
> s=Asterisk PBX 1.6.1.13
> c=IN IP4 10.153.66.146
> t=0 0
> m=image 4819 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxFillBitRemoval
> a=T38FaxTranscodingMMR
> a=T38FaxTranscodingJBIG
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxDatagram:1400
> a=T38FaxUdpEC:t38UDPRedundancy
> 
> I'm pretty sure it doesn't support T38FaxTranscodingMMR or 
> T38FaxTranscodingJBIG, so they should not be there. Perhaps more 
> relevant to you, though, is why is * saying "(External RTP bridge)". 
> Does it really mean it?
> 
> Steve

I'm not really sure. What I know is that this telco has separate boxes for SIP signalling and RTP media. Not even sure if that's related to your question which, to be honest, I didn't fully understand. :)



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