[Asterisk-Users] fax pass thru issue

Tue Mar 16 13:49:29 MST 2004

Gurus,

I currently have two separate asterisk networks running but am having
problems on one and not the other with identical configs (other than
transport, one if over the internet one is a managed network, ironically the
over the Internet pass thru is working).

I have both systems configured identical, the gateways and sipuras are
configured identical as well, outside of network parameters like IPs, etc.

On the network that passes over the Internet I dial from a fax machine
connected to the sipura unit , the call goes through the asterisk box, and
then to the gateway to the pstn. I had problems initially when using rfc2833
as my dtmf selection on the sipura to asterisk side with inband from
asterisk to tnt , the calls would get a fast busy if dialing from a fax, but
would work as voice calls.

I then changed everything to inband, and internal dtmf (sipura to asterisk)
would work and dtmf would work on external calls to the pstn, and fax would
pass. So i thought I had success.

I then configured the second system which goes over a managed backbone (to
be clear this problem is not a network issue, i just want to be clear on the
differences), the configs are identical but I get a fast busy when making a
fax call from the sipura through the asterisk unit, but if I make a voice
call on the same line it works.

The major thing I notice in the debugs is that it seems to be treating the
fax tones as dtmf digits and then says it cannot pass an f...........

I have put some info on the scenario below as well as the debugs. I have
changed #s and IPs to protect the innocent:) All help is truly appreciated.

Anyway, here is the scenario:

Fax machine ---> Sipura ----> IP Network ---> Asterisk ----> Lucent MAX TNT
with SIP ---> PSTN to FAX

Here is a sample user config on asterisk for the sipura unit:
(interchange a valid area code for 555)

[5554377199]
disallow=all
allow=ulaw
type=friend
username=5554377199
secret=XXXXX
host=dynamic
context=sipcalls
canreinvite=no
dtmfmode=inband
nat=1

And here is the sip.conf for the connection to the MAX:

[general]
port= 5060
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
context = sipcalls

[XXXXsip]
context=XXXXusersincoming
type=friend
host=XXX.201.XXX.157
dtmfmode=inband
disallow=all
allow=ulaw

#include "/etc/asterisk/sipusers.conf"

    -- Registered SIP '5554376785' at 55.214.555.146 port 50035 expires 3600
Mar 16 08:21:29 DEBUG[1125329600]: chan_sip.c:4220 check_user: Setting NAT
on RTP to -1
Mar 16 08:21:29 DEBUG[1125329600]: chan_sip.c:598 __sip_ack: Stopping
retransmission on '9e3843a7-1867c6ae at 192.168.23.53' of Response 101: Found
Mar 16 08:21:29 DEBUG[1125329600]: chan_sip.c:4220 check_user: Setting NAT
on RTP to -1
Mar 16 08:21:29 DEBUG[1125329600]: chan_sip.c:5318 handle_request: Check for
res for 5554376785
Mar 16 08:21:29 DEBUG[1125329600]: chan_sip.c:1150 find_user: Call from user
'5554376785' is 1 out of 0
Mar 16 08:21:29 DEBUG[1125329600]: chan_sip.c:3596 build_route: build_route:
Contact hop: Warmington Villa Fax <sip:5554376785 at 192.168.23.53:5060>
    -- Executing Dial("SIP/5554376785-fffb", "SIP/15552468901 at XXXsip|200")
in new stack
Mar 16 08:21:29 DEBUG[1225991360]: chan_sip.c:818 create_addr: Setting NAT
on RTP to 0
Mar 16 08:21:29 DEBUG[1225991360]: chan_sip.c:1023 sip_call: Outgoing Call
for 15552468901
Mar 16 08:21:29 DEBUG[1225991360]: chan_sip.c:1122 find_user: 15552468901 is
not a local user
    -- Called 15552468901 at XXXsip
Mar 16 08:21:29 DEBUG[1125329600]: chan_sip.c:618 __sip_semi_ack:
(Provisional) Stopping retransmission (but retaining packet) on
'7d0800c334ad1ac85714e9c951d6934a at 55.201.555.158' Request 102: Found
Mar 16 08:21:29 DEBUG[1125329600]: chan_sip.c:618 __sip_semi_ack:
(Provisional) Stopping retransmission (but retaining packet) on
'7d0800c334ad1ac85714e9c951d6934a at 55.201.555.158' Request 102: Found
    -- SIP/xxxsip-dc3a is making progress passing it to SIP/5554376785-fffb
Mar 16 08:21:29 DEBUG[1225991360]: rtp.c:1015 ast_rtp_write: Ooh, format
changed from UNKN to ULAW
Mar 16 08:21:29 DEBUG[1225991360]: rtp.c:389 ast_rtp_read: RTP NAT: Using
address 55.214.555.146:50121
Mar 16 08:21:29 DEBUG[1225991360]: rtp.c:1015 ast_rtp_write: Ooh, format
changed from UNKN to ULAW
Mar 16 08:21:29 DEBUG[1125329600]: chan_sip.c:618 __sip_semi_ack:
(Provisional) Stopping retransmission (but retaining packet) on
'7d0800c334ad1ac85714e9c951d6934a at 55.201.555.158' Request 102: Found
    -- SIP/xxxsip-dc3a is ringing
Mar 16 08:21:29 DEBUG[1225991360]: channel.c:1228 ast_indicate: Driver for
channel 'SIP/5554376785-fffb' does not support indication 3, emulating it
Mar 16 08:21:29 DEBUG[1225991360]: channel.c:1337 ast_prod: Prodding channel
'SIP/5554376785-fffb'
Mar 16 08:21:30 DEBUG[1125329600]: chan_sip.c:580 __sip_ack: Acked pending
invite 102
Mar 16 08:21:30 DEBUG[1125329600]: chan_sip.c:598 __sip_ack: Stopping
retransmission on '7d0800c334ad1ac85714e9c951d6934a at 55.201.555.158' of
Request 102: Found
Mar 16 08:21:30 DEBUG[1125329600]: chan_sip.c:3596 build_route: build_route:
Contact hop: <sip:16502468901 at 64.201.123.157:5060;user=phone>
    -- SIP/xxxsip-dc3a answered SIP/5554376785-fffb
    -- Attempting native bridge of SIP/5554376785-fffb and SIP/racesip-dc3a
Mar 16 08:21:30 DEBUG[1125329600]: chan_sip.c:598 __sip_ack: Stopping
retransmission on '9e3843a7-1867c6ae at 192.168.23.53' of Response 102: Found
Mar 16 08:21:30 DEBUG[1225991360]: rtp.c:950 ast_rtp_raw_write: Difference
is 4016, ms is 522
Mar 16 08:21:30 DEBUG[1225991360]: rtp.c:950 ast_rtp_raw_write: Difference
is 4144, ms is 538
Mar 16 08:22:22 DEBUG[1225991360]: channel.c:2304 ast_channel_bridge: Didn't
get a frame from channel: SIP/5554376785-fffb
Mar 16 08:22:22 DEBUG[1225991360]: channel.c:2372 ast_channel_bridge: Bridge
stops bridging channels SIP/5554376785-fffb and SIP/xxxsip-dc3a
Mar 16 08:22:22 DEBUG[1225991360]: chan_sip.c:1226 sip_hangup:
find_user(15552468901) - decrement outUse counter
Mar 16 08:22:22 DEBUG[1225991360]: chan_sip.c:1122 find_user: 15552468901 is
not a local user
  == Spawn extension (sipcalls, 15552468901, 1) exited non-zero on
'SIP/5554376785-fffb'
    -- Executing Hangup("SIP/5554376785-fffb", "") in new stack
  == Spawn extension (sipcalls, h, 1) exited non-zero on
'SIP/5554376785-fffb'
Mar 16 08:22:22 DEBUG[1225991360]: chan_sip.c:1229 sip_hangup:
find_user(5554376785) - decrement inUse counter
Mar 16 08:22:22 DEBUG[1125329600]: chan_sip.c:598 __sip_ack: Stopping
retransmission on '7d0800c334ad1ac85714e9c951d6934a at 55.201.555.158' of
Request 103: Found
localhost*CLI>