gmane.comp.telephony.pbx.asterisk.devel

Warning message regarding t38

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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Denis Galvão | 14 Aug 22:06

Re: Warning message regarding t38

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!
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev

_______________________________________________
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Johan Wilfer | 14 Aug 22:47

Re: Warning message regarding t38

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!
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>

_______________________________________________
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