[asterisk-dev] Warning message regarding t38

Thiago Fernandes Thiago.Fernandes at locaweb.com.br
Thu Aug 14 13:24:49 CDT 2008


We have a Asterisk 1.4.17 set to receive calls from a SIP provider. This
company sends a INVITE directive which is new for me. It encapsulates
some T38 parameters, which make Asterisk fire a warning message, as
shown below:
 
WARNING[31937]: chan_sip.c:5083 process_sdp: Unsupported SDP media type
in offer: image 58748 udptl t38

Please see the complete INVITE directive:
<--- SIP read from 1X.1XX.1XX.208:5060 --->
INVITE sip:1130413900 at 1X.2XX.137.50:5060;transport=UDP;user=phone
SIP/2.0
f:
<sip:1130799781 at 1X.1XX.1XX.208:5060;user=phone>;tag=c0a-13c4-10e56f-53c6
14b4-10e56f
t: <sip:1130413900 at 1X.2XX.1XX.50:5060;user=phone>
i: a15d7878d0838c0a13c410e56f42003c39eb3e0440c23d6f8-0542-4635
CSeq: 1 INVITE
User-agent: CS2000_NGSS/9.0
P-Asserted-Identity: <sip:1130799781 at 1X.1XX.1XX.208;user=phone>
Max-Forwards: 140
k: 100rel
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP
CTA1CS2K:5060;maddr=10.140.131.208;branch=z9hG4bK-10e56f-42003c39-5dc0e7
f6
m: <sip:1X.1XX.1XX.208:5060;transport=UDP>
c: application/SDP
l: 414
v=0
o=PVG 1218715242070 1218715242070 IN IP4 10.142.1.89
s=-
p=+1 6135555555
c=IN IP4 10.142.1.89
t=0 0
m=audio 50556 RTP/AVP 18 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=fmtp:18 annexb=no
m=image 58748 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
 
<------------->
--- (14 headers 18 lines) ---
Sending to 1X.1XX.1XX.208 : 5060 (no NAT)
Using INVITE request as basis request -
a15d7878d0838c0a13c410e56f42003c39eb3e0440c23d6f8-0542-4635
Found peer 'GVTTRK01'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
[Aug 14 12:00:22] WARNING[31937]: chan_sip.c:5083 process_sdp:
Unsupported SDP media type in offer: image 58748 udptl t38
Peer audio RTP is at port 10.142.1.89:50556
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108
(alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 1X.1XX.1.89:50556
Looking for 1130413900 in incoming_voxip (domain 10.213.137.50)
list_route: hop: <sip:1X.1XX.1XX.208:5060;transport=UDP>
 
 
On the other end of this call, we have a user agent which only accepts
audio, via codecs such as G711, G729 and GSM. This is fine, because we
actually do not want to use FAX now, only want a regular audio session.
However, Asterisk keeps showing that warning message everytime a new
call arrives, which is quite annoying.
 
We tried to enable SIP parameter t38pt_udptl to yes. In fact, the
warning message has gone after that, but we started to get the below
error and the call is hang:
 
ERROR[31937]: chan_sip.c:12242 handle_response_invite: Got error on T.38
initial invite. Bailing out.
 
How do we get rid of that warning message, without enabling t38pt_udptl
to yes?
 
Thanks in advance!
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