[asterisk-dev] Warning message regarding t38

[Denis Galvão]

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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