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

[Robin Inderberg]

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> 

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