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Fax pass-through with ATA 186

x.thomas
Level 1
Level 1

Hello,

I need to use Fax with ATA 186. I want to use "Fax pass-through mode". I updated my ATA with the last firmware for SCCP.

I can send and receive faxes when I use G711 codecs but when I use codec G729, it fails.

I configured my gateway :

o voice service voip

o modem passthrough NSE codec g711alaw redundancy 5

o fax t38 inhibit

On my ATA I put 90000480 for connect mode parameter and 0x00150015 for audiomode.

What is the problem ??? Thanks for your help.

I join logs obtained with prserv.

--------------------------------------------------

|->CallState[0]=5 L=1 R=1000137 FG=0 r=1000137,0

|->CallInfo[0]

Save Media[0]

|->StartMediaTx[0]:pt=18 ps=20 ss=0 ip=c0a8c805:18452

[0]MPT mode 0

[2]MPT mode 0

SetG723Mode: 2 3

Enable LEC adapt [0]=1

[0]Enable encoder 18

[0]DPKT 1st: 28369 28129, pt 18

CED

[0]Tx MPT PT=100 NSE pkt c0000000

[0]MPT mode 2

[0]codec: 18 => 0

7:00;6,0,0,0,

[0]Tx MPT PT=100 NSE pkt c0000000

[0]codec: 0 => 0

[0]Tx MPT PT=100 NSE pkt c0000000

[0]codec: 0 => 0

[1]StopTone

[1]Connecting to <c0a8c80a:2000> <-1:25>...

[1]<Conn:c0a8c80a soc:26> ok!

REG:ATA0CCE91C2A601

|->RegRej[1]:Error: DB Config

[1]StopTone

[1]boxreset<8:1>

7:30;6,0,0,0,

[0]Tx MPT PT=100 NSE pkt c0000000

[0]codec: 0 => 0

[0]Tx MPT PT=100 NSE pkt c0000000

[0]codec: 0 => 0

<-|EndCall[0]

<-|SK[0]:9, L=1, FGR=1000137

|->CallState[0]=2 L=1 R=1000137 FG=0 r=1000137,0

IDLE[0]

->CloseRxChn[0]

|->StopMediaTx[0]

|->SetRingerMode[0]:1

[1]StopTone

[1]Connecting to <c0a8c80a:2000> <-1:26>...

[1]<Conn:c0a8c80a soc:27> ok!

REG:ATA0CCE91C2A601

|->RegRej[1]:Error: DB Config

[1]StopTone

[1]boxreset<9:1>

8:00;6,0,0,0,

8:30;6,0,0,0,

[1]StopTone

[1]Connecting to <c0a8c80a:2000> <-1:27>...

[1]<Conn:c0a8c80a soc:28> ok!

REG:ATA0CCE91C2A601

|->RegRej[1]:Error: DB Config

[1]StopTone

[1]boxreset<10:1>

9:00;6,0,0,0,

----------------------------------------------------

Thanks for your help...

4 Replies 4

Aaron Harrison
VIP Alumni
VIP Alumni

Hi

The only mistake I see is that you have your gateway set to g711alaw for upspeed... Your ATA is configured for g711ulaw.

Either change your gateway, or use this connectmode for the ATA:

0x90002480

Have you configured your GW as MGCP or H323?

Aaron

Aaron Please remember to rate helpful posts to identify useful responses, and mark 'Answered' if appropriate!

Hi,

I have ever tried with "connect mode = 0x9002480" but I have again the problem. I want to work with g711alaw so I keep "connect mode = 0x9002480".

My gateway is configured as MGCP with the exterior but as SCCP with my ATA.

Do you tink there is a problem with that ?

Thanks

Help !!!

I have not resolved my problem...

Is there somebody that can help me ????

Thanks in advance.

Hi Koushik!

We use overlay ephone-dn for the following scenario

Every phone has one dedicated ephone-dn with the interal number (and the external) as secondary.

There is a "group"number which should ring on every phone.

To put all together we use overlay ephone-dn.

This works also quit good.

But here comes the problem:

When someone calls the groupnumber, every phone rings.

Someone answers the phone. -> This works fine.

Now someone else calls the dedicated phone number of this phone which is in call.

This second call does not receive busy signal but just normal ringing.

The phoneuser does not see that a second call is being made to him/her(as it would be if we use a 2 button model (button 1 internal, button 2 groupnumber(shared)).

If the phoneuser hangs up the first call, then the ring of the second call is seen to the phone user.

This is not what we want. If a phone is busy, it should not be possibile to make a call to the phone as long as it is busy.

To make it a bit more clear here the (basic) config

ephone-dn 1

number 1 secondary 11

ephone-dn 2

number 2 secondary 22

ephone-dn 3

nunber 33

ephone-dn 4

number 33

ephone 1

button 1o1,3,4

ephone 2

button 1o2,3,4

Someone calls 33

Ephone 1 ansers

Someone calls 11 (or 1) and hears normal ringing instead of busy, since ephone 1 is speeching

To bit more precise we use dualline (but with the huntstop channel feature) to give the possibility of transfer with consult. We tested also with every ephone-dn not being dualline, same result

Thanks a lot for any hint oder help

Paul Knoll