[asterisk-bugs] [Asterisk 0012437]: Asterisk negotiates only T.38 when answering even if the other end offers audio

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12437 
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Reported By:                marsosa
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   12437
Category:                   Channels/chan_sip/T.38
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.4.18 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-04-14 08:31 CDT
Last Modified:              2009-03-03 17:48 CST
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Summary:                    Asterisk negotiates only T.38 when answering even if
the other end offers audio
Description: 
One of our gateways (audiocodes mp-118) offers ulaw,g729 and t.38 when an
incoming call is sent to asterisk, and asterisk answer() with t.38 only,
instead of using ulaw. T.38 is enabled on the gateway because this is
needed for reinvites, if i disable it, the call works ok but fails later
when the ata wants to do reinvite for receiving faxes with t.38 '488 not
acceptable'.
The main problem here is that, after answering with t.38, asterisk sends
invites with t.38 only to the ip phones, and they rejected with not
acceptable.
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 (0101161) pinga-fogo (reporter) - 2009-03-03 17:48
 http://bugs.digium.com/view.php?id=12437#c101161 
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I'll try this branch and sometings works sometings not... 

Scenario 1

A Call from nortel mcs(always send t38 in the inicial invite)to asterisk,
in the dialplan i do a Answer and play a audio file.. this works fine...the
asterisk offers audio...

MCS ---> Asterisk --> file playback

Scenario 2

MCS ---> Asterisk --> Poycom Ip430
This scenario does not work, the 

in the sip debug i see the codec negotiations is ok
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x108
(alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
but after the dial command the asterisk shows
   -- Executing [s at macro-stdexten:2] Dial("SIP/ip4440015253f-0070d2d0",
"SIP/5002&SIP/5008|30|tTj") in new stack
[Mar  3 14:07:46] WARNING[16420]: chan_sip.c:3081 sip_call: No audio
format found to offer. Cancelling call to 5002
    -- Couldn't call 5002
[Mar  3 14:07:46] WARNING[16420]: chan_sip.c:3081 sip_call: No audio
format found to offer. Cancelling call to 5008
    -- Couldn't call 5008

and the 5002 and 5008 is ok .. all are the codecs 729 and 711 enabled and
configured

sorry about my english ... 
i hope this can help us 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-03 17:48 pinga-fogo     Note Added: 0101161                          
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