[asterisk-users] Has anyone used FaxGateway()

[James Lamanna]

Hi,
Here's part of the log that I see.
In this case I'm testing on a box that unfortunately doesn't have a
PRI connection.
I've so far tested with just voice calls so far, but as you can see,
FaxGateway can't even dial out to the SIP trunk properly.

Here's also what the dialplan looks like:

exten => _1NXXNXXXXXX,1(faxtest),FaxGateway(SIP/vitel-outbound/${EXTEN},-1)

And the log:

[Jan 14 21:04:04] VERBOSE[8110] logger.c:     -- Executing
[1xxxxxxxxxx at from-internal:14] FaxGateway("SIP/xxxxxxxxxx-098befd0",
"SIP/vitel-outbound/1xxxxxxxxxx|-1") in new stack
[Jan 14 21:04:04] WARNING[8110] rtp.c: Unable to set TOS to 184
[Jan 14 21:04:04] WARNING[8110] udptl.c: UDPTL unable to set TOS to 184
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - transmit entry.
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: FaxGw - transmit -
type SIP destination vitel-outbound/1xxxxxxxxxx
[Jan 14 21:04:04] VERBOSE[8110] logger.c: Called vitel-outbound/1xxxxxxxxxx
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - after ast_call.
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - waiting for
activity on channels
[Jan 14 21:04:04] ERROR[7258] chan_sip.c: Got error on T.38 initial
invite. Bailing out.
[Jan 14 21:04:04] DEBUG[7258] chan_sip.c: change_t38_state chnaged state to: 0
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - something
happend on peer
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - AST_FRAME_CONTROL
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - AST_CONTROL_BUSY
[Jan 14 21:04:04] DEBUG[8110] app_faxgateway.c: faxGw - connections
build - ready 0 and erady to talk - 1
[Jan 14 21:04:04] ERROR[8110] app_faxgateway.c: failed to get
remote_channel SIP vitel-outbound/1xxxxxxxxxx
[Jan 14 21:04:04] NOTICE[8110] app_faxgateway.c: FaxGateway exit with CONGESTION
[Jan 14 21:04:04] WARNING[8110] app_faxgateway.c: Transmission error