Issue: I am attempting to send Fax’s via the FACsys software using the SoftIP software. I am able to make the call, it appears to connect, but the fax fails with a status of “Remote unit incompatible” in the FACSys Desktop software. Below I have included the output of “sip show channelstats” because it appears that packets are being lost.
SIP GLOBAL includes:
T.38 support: Yes
T.38 EC mode: Redundancy
T.38 MaxDtgrm: -1
I have a Linsys ATA with a fax machine behind a NAT that connects to the same Asterisk box and it is able to send faxes to the destination which the FACSys server cannot.
Setup:
Fax Server
*internal IP behind NAT
*SoftIP Software (dialogic.com/products/ip_ena … re_SIP.htm).
*FACsys faxing software
*connects via SIP proxy to asterisk server
Asterisk Server
*1.8.4-rc2
*behind NAT
sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
<providerIP> 787a8afe620 00:00:18 0000000166 0000000007 ( 4.05%) 0.0000 0000000169 0016776815 (9927109.47%) 0.0002
<externalIP of FACSys Server> da929008-15 00:00:18 0000000169 0000000000 ( 0.00%) 0.0000 0000000164 0000000000 ( 0.00%) 0.0145
2 active SIP channelsSIP flow from packet capture:
Conv.| Time | <SoftIP ExternIP> | <SIP Provider IP> |
| | | <asterisk internIP> |
0 |30.821 | INVITE SDP (g711U telephone-eventRTPType-101) | |SIP From: "SoftIP" <sip:softip@server To:sip:<dialed#>@asteriskbox.com
| |(5060) ------------------> (5060) | |
0 |30.821 | 100 Trying| | |SIP Status
| |(5060) <------------------ (5060) | |
-----------------------------------------------------------------------------
1 |30.830 | | INVITE SDP (g711U g711A telephone-eventRTPType...1) |SIP From: "SoftIP" <sip:softip@<server internIP> To:<sip:<dialed#>@provider.com
| | |(5060) ------------------> (5060) |
1 |30.909 | | 100 Trying| |SIP Status
| | |(5060) <------------------ (5060) |
1 |38.305 | | 183 Session Progress SDP (g711U telephone-even...PType-101) |SIP Status
| | |(5060) <------------------ (5060) |
-----------------------------------------------------------------------------
0 |38.306 | 183 Session Progress SDP (g711U telephone-even...PType-101) | |SIP Status
| |(5060) <------------------ (5060) | |
-----------------------------------------------------------------------------
1 |41.724 | | 200 OK SDP (g711U telephone-eventRTPType-101) |SIP Status
| | |(5060) <------------------ (5060) |
1 |41.724 | | ACK | |SIP Request
| | |(5060) ------------------> (5060) |
-----------------------------------------------------------------------------
0 |41.725 | 200 OK SDP (g711U telephone-eventRTPType-101) | |SIP Status
| |(5060) <------------------ (5060) | |
0 |41.747 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
-----------------------------------------------------------------------------
1 |83.103 | | BYE | |SIP Request
| | |(5060) <------------------ (5060) |
1 |83.103 | | 200 OK | |SIP Status
| | |(5060) ------------------> (5060) |
-----------------------------------------------------------------------------
0 |83.104 | BYE | | |SIP Request
| |(5060) <------------------ (5060) | |
0 |83.121 | 200 OK | | |SIP Status
| |(5060) ------------------> (5060) | |rtp set debug on
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029030, ts 2896991065, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029031, ts 2896991225, len 000160)
-- SIP/<PROVIDER-SIP-DNS>.com-00000031 is making progress passing it to SIP/<SOFTIP-HOST>-00000030
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029032, ts 2896991385, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027479, ts 2896991384, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029033, ts 2896991545, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027480, ts 2896991544, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029034, ts 2896991705, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027481, ts 2896991704, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029035, ts 2896991865, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027482, ts 2896991864, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029036, ts 2896992025, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027483, ts 2896992024, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029037, ts 2896992185, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027484, ts 2896992184, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029038, ts 2896992345, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027485, ts 2896992344, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029039, ts 2896992505, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027486, ts 2896992504, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029040, ts 2896992665, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027487, ts 2896992664, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029041, ts 2896992825, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027488, ts 2896992824, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029042, ts 2896992985, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027489, ts 2896992984, len 000160)
Got RTP packet from <PROVIDER-IP>:21260 (type 00, seq 029043, ts 2896993145, len 000160)
Sent RTP packet to <SOFTIP-EXTERNIP>:10052 (type 00, seq 027490, ts 2896993144, len 000160)
-- SIP/<PROVIDER-SIP-DNS>.com-00000031 answered SIP/<SOFTIPHOST>-00000030
after the call is answered it appears that RTP stops. Maybe due to the face that canreinvite = yes.
changed sip.conf canreinvite = no
[2011-03-16 00:26:27] DEBUG[13222]: res_rtp_asterisk.c:1437 process_dtmf_rfc2833: Ignoring RTP 2833 Event: 00000024. Not a DTMF Digit.
[2011-03-16 00:26:27] DEBUG[13222]: res_rtp_asterisk.c:1437 process_dtmf_rfc2833: Ignoring RTP 2833 Event: 00000024. Not a DTMF Digit.
[2011-03-16 00:26:27] DEBUG[13222]: res_rtp_asterisk.c:1437 process_dtmf_rfc2833: Ignoring RTP 2833 Event: 00000024. Not a DTMF Digit.
asteriskbox*CLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
<PROVIDER-IP> 42d8edb81f5 00:00:51 0000002240 0000000003 ( 0.13%) 0.0000 0000002235 0000000000 ( 0.00%) 0.0013
<SOFTIP-EXTERNIP> efd6f438-10 00:00:51 0000002701 0000003435 (55.98%) 0.0000 0000002239 0000000000 ( 0.00%) 0.0154
2 active SIP channels
Does anyone have any idea what may be causing this?
Thanks in advance,
Rick