[asterisk-dev] Warning message regarding t38

Thu Aug 14 15:47:15 CDT 2008

I think the patch at bug #12746 solves your problem.
http://bugs.digium.com/file_download.php?file_id=19341&type=bug
I had the same sideeffekt as you report, but my main issue was the
broken calls. After I applied the patch the calls didn't
break any more nor did I get those anoying messages...

Good luck!
/Johan

2008/8/14 Denis Galvão <denisgalvao at gmail.com>:
> Hi Thiago.
>
> Is your sip peer configured with canreinvite=yes?
>
> Disable it.
>
> --
> Denis Galvão
> AsteriskBrasil.org
>
> Ajude a comunidade AsteriskBrasil.org, compre uma camiseta!
> http://www.voipmania.com.br
>
>
> On 14/08/2008, at 15:24, Thiago Fernandes wrote:
>
>> 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-53c614b4-10e56f
>> 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-42003c39-5dc0e7f6
>> 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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