[asterisk-bugs] [Asterisk 0011066]: Asterisk SIP Connections to systems that support t38 fax detection may fail

[noreply at bugs.digium.com]

The following issue has been RESOLVED. 
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http://bugs.digium.com/view.php?id=11066 
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Reported By:                jon
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   11066
Category:                   Channels/chan_sip/T.38
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     resolved
Asterisk Version:            1.4.12  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 not fixable
Fixed in Version:           
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Date Submitted:             10-23-2007 14:06 CDT
Last Modified:              03-19-2008 12:26 CDT
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Summary:                    Asterisk SIP Connections to systems that support t38
fax detection may fail
Description: 
I've got a list of about 5 Direct Voice or IVR numbers right now that we
can not call using asterisk SIP. 

Using SIP, the following errors are logged (full debug attached in
sip-debug.txt):

[Oct 22 14:49:14] WARNING[6327]: chan_sip.c:5017 process_sdp: Unsupported
SDP me
dia type in offer: image 19596 udptl t38
    -- SIP/bandwidth-08444138 is making progress passing it to
SIP/jon-b5e00468
[Oct 22 14:49:24] WARNING[6327]: chan_sip.c:5017 process_sdp: Unsupported
SDP me
dia type in offer: image 19596 udptl t38
    -- SIP/bandwidth-08444138 answered SIP/jon-b5e00468
  == Spawn extension (local, dial-12128482036, 6) exited non-zero on
'SIP/ingramoffice2-b5e00468'
[Oct 22 14:49:26] WARNING[6327]: chan_sip.c:12528 handle_response: Remote
host can't match request BYE to call
'4944de5b1fc061d90823411753ee8fc4 at 4.68.250.148'. Giving up.


Calling with IAX2 and ZAP works fine, the numbers ring and the IVR or user
answers.
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Relationships       ID      Summary
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related to          0008078 T38 relay doesn't work between Audiocod...
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---------------------------------------------------------------------- 
 file - 03-19-08 12:26  
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I've researched this and come to the conclusion that there is currently
nothing in chan_sip we can do to accomodate this. We send out an initial
INVITE with audio, but instead of getting audio in the 200 OK the remote
side sends T38. We can't "fall back" so we terminate the call. 

Issue History 
Date Modified   Username       Field                    Change               
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03-19-08 12:26  file           Note Added: 0084257                          
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