[asterisk-users] Trouble with T38/Dialogic

[Andrew McRory]

Hello, I am working with 1.8.32.2 which I have patched with t38-gateway and
PRACK. t38 is tested and working fine with Zoiper client but I can't get the
t.38 software from Biscom (FAXCOM) to talk. In my first attempts I found
FAXCOM announces that it supports 100rel so I added the PRACK patch hoping
that would do the trick. Now it gets a little further but * complains about
rejecting a non-primary audio stream.

Could this be a problem with 1.8 not liking the second media stream or is
there some more configuration tweaking to be done?

--- CUT ----------------------------------------------
<--- SIP read from UDP:192.168.1.13:5060 --->
INVITE sip:1XXXXXXXXXX at 192.168.1.11 SIP/2.0
From: Biscom
<sip:418 at 192.168.1.13>;tag=86c9140-d281eac-13c4-55013-1f571-33180d4a-1f571
To: <sip:1XXXXXXXXXX at 192.168.1.11>
Call-ID: 73cb2e8-d281eac-13c4-55013-1f571-7ef285e4-1f571
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-1f571-7a6c245-2c788e59
Supported: 100rel
Max-Forwards: 70
User-Agent: Brktsip/6.6.9B12 (Dialogic)
Contact: <sip:192.168.1.13>
Authorization: Digest
username="418",realm="asterisk",nonce="37a417d1",uri="sip:1XXXXXXXXXX at 192.168.1.11",response="6cdb70491bdaf9acc75d9b776101d111",algorithm=MD5
Content-Type: application/sdp
Content-Length: 220

v=0
o=418 2209120086 0667748000 IN IP4 192.168.1.13
s=no_session_name
t=0 0
m=audio 56040 RTP/AVP 0
c=IN IP4 192.168.1.13
a=rtpmap:0 pcmu/8000
m=audio 56040 RTP/AVP 8
c=IN IP4 192.168.1.13
a=rtpmap:8 pcma/8000
<------------->
--- (13 headers 10 lines) ---
Sending to 192.168.1.13:5060 (no NAT)
Using INVITE request as basis request -
73cb2e8-d281eac-13c4-55013-1f571-7ef285e4-1f571
Found peer '418' for '418' from 192.168.1.13:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found audio description format pcmu for ID 0
[2015-02-12 23:36:58] WARNING[13727]: chan_sip.c:9305 process_sdp: Rejecting
non-primary audio stream: audio 56040 RTP/AVP 8

<--- Reliably Transmitting (no NAT) to 192.168.1.13:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.1.13:5060;branch=z9hG4bK-1f571-7a6c245-2c788e59;received=192.168.1.13
From: Biscom
<sip:418 at 192.168.1.13>;tag=86c9140-d281eac-13c4-55013-1f571-33180d4a-1f571
To: <sip:1XXXXXXXXXX at 192.168.1.11>;tag=as7bee4e61
Call-ID: 73cb2e8-d281eac-13c4-55013-1f571-7ef285e4-1f571
CSeq: 2 INVITE
Server: FPBX-12.0.37(1.8.32.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, PRACK, MESSAGE
Supported: replaces, timer, 100rel
Content-Length: 0
--- CUT ----------------------------------------------

TIA

--
Andrew McRory
Sayso Communications, Inc.
Tallahassee, Florida