[cisco-voip] ATA186 sccp through a SIP trunk to TELCO

[Robin Inderberg]

Hi,

 

Thank your for taking time to answer my question Tim.

 

This seems like it could be right as you say, often it comes after 3-4 pages, and even if 3-4 pages come through and I get communication error and the transmission is broke. The first page could look really bad. 

 

Is it any solutions for this problem? Or is it hopeless? It must be possible to solve I guess, otherless no one should use the ata with fax machines at all. And what I know, people have got this working? 

 

If you could think of any solution of this, please tell me. 

 

Thank you once again, 

 

Med vänliga hälsningar

 

Robin Inderberg

robin.inderberg at candidator.se <mailto:robin.inderberg at candidator.se> 

_____________________________________________

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________________________________

From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
Sent: den 23 augusti 2006 16:14
To: Robin Inderberg
Subject: RE: [cisco-voip] ATA186 sccp through a SIP trunk to TELCO

 

This sounds like the fax sync tone 'slippage' for Group III fax machines...

 

I recall an issue that Cisco discussed some years ago where the fax sync tones between two machines would stay in sync for the first couple of pages, but they were moving out of sync with each other slowly.

After about 3-4 pages, the  tones might not match up at all, and the fax would error out..

 

Best I can recall, anyway..

 

There was a CD that Cisco had given us partners (I was a partner back then, now I'm an end user) that had some discussions of how this worked..

Something about the delays inherent in digital lines not being 'nice' to the sync tones..

 

 

I may be horribly out of date on my thinking here--- Wes Sisk can correct me, I'm sure...

 

T

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robin Inderberg
Sent: Wednesday, August 23, 2006 8:48 AM
To: Cisco Voip
Subject: [cisco-voip] ATA186 sccp through a SIP trunk to TELCO

Hi,

 

We connect to the PSTN through a SIP trunk to our Telco, they do have a Cisco SIP Proxy Server.

And we have Ata 186 with fax passthrough on our side, when we succeed with one page, we only get the top of the page, and the page are cut there. 

 

And when we send, we get communication error on the faxes. That doesn't say very much at all. 

 

We use the following audio mode and connect mode. 

AudioMode 0x00150015

ConnectMode 0x90002400

 

Our telco use the following settings, 

 

voice class codec 5035

 codec preference 1 g729br8

 codec preference 2 g711alaw

 codec preference 3 g711ulaw

 

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw 

 

        Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122

               CAS=123, ClearChan=125, PCM switch over u-law=0,A-law=8

               G726r16 using static payload

               G726r24 using static payload

->        fax rate = fax,   payload size =  20 bytes

->        fax protocol = t38

->        fax-relay ecm enable

->        fax NSF = 0xAD0051 (default)

->        codec = g729r8,   payload size =  20 bytes,

->        Media Setting = flow-through (global)

->        Expect factor = 10, Icpif = 20,

->        Playout Mode is set to adaptive,

->        Initial 60 ms, Max 250 ms

 

Do you have any idea of what could be wrong? 

 

Med vänliga hälsningar

 

Robin Inderberg

robin.inderberg at candidator.se <mailto:robin.inderberg at candidator.se> 

_____________________________________________

Telefon:   0322-67 10 00    www.candidator.se <http://www.candidator.se/> 

Direkt:    0322-67 10 19    Candidator AB

Mobil:     0733-47 10 19    Malmgatan 15

Fax:       0322-67 10 99    441 39  ALINGSÅS

¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯

 

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