I have 4 ATA 186 SCCP, registed in CCM4.1(2).
I have configured my h323 gateway for fax passthrough.
I leave default parameters in audio and connect mode.
In line 1 of ATA i don't have problems. but in line 2 i can't receive fax from PSTN, i tested with an internal fax and i have received fax.
It isn't a dial-peer problem because if configure this DN in line 1 of ata it works.
In line 2 of ATA fax machine ring once and then calling party ear busy tone. changing fax machine by an analog phone, phone rings but when i answer call i ear busy tone.
Could someone give me an help?
There is a caveat with the 188 and the fact that the unit can only handle one G729 call at any one time. The second line will fall back to G711. Not sure if this is the problem but is good to know.
I'd suggest you change the audiomode on the ATA to be 0x00120012 to make both lines default to G711. We had real issues with faxes until we implemented this along with a couple of other changes (E1 clock etc).
Just re-read your message. I beleive Cisco don't suggest using ATA 186's to do faxing. There's a couple of conflicting articles out there, but faxing isn't really supported with the 186 devices. 188's are kind of supported AFAIK.
I have never heard that ATA 186 I2 didn't support faxing.
Could you send me please where did you saw that info?
See this note for the fax issue referred to above - the I2 ATA variant is OK, it's the older one...
To get ATA working with passthru see this guide, you need to change the audiomode paramater (i think, from memory, that the connectmode specified here is the default setting):
My ATA , I2 ATA.
My audio mode is in ATA is 0X00350035 (default) , and line 1 in ATA is working, line 2 only sends fax.
So, i have to change the config file in CCM because i'm receiving config from TFTP server.? or can i just change audio mode in config web page of ATA?
For clarification, you are saying that line 2 can actually send faxes successfully, you only have problems when that ATA port tries to receive a call, either a fax call, or a call to an attached phone? Maybe you could go on the H323 gateway when things are quiet and do a 'debug voip ccapi inout' and place a test call in to the problem port, and see what your call is cleared back to..
And, to follow on to the second post about the codec issue, if you don't specify a codec on your dial peer, it will want to use g729 as default, so you might try specifying g711 for a test to see if codec is your problem, or you can do a voice class codec config to allow a few codecs in order of preference, and then apply that to your dial peer