[asterisk-bugs] [Asterisk 0016107]: t38 passtrough not working with kapanga softphone and eutelia provider

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Oct 22 16:11:02 CDT 2009


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16107 
====================================================================== 
Reported By:                darkbasic
Assigned To:                kpfleming
====================================================================== 
Project:                    Asterisk
Issue ID:                   16107
Category:                   Channels/chan_sip/T.38
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2 
SVN Revision (number only!): 224859 
Request Review:              
====================================================================== 
Date Submitted:             2009-10-20 18:12 CDT
Last Modified:              2009-10-22 16:11 CDT
====================================================================== 
Summary:                    t38 passtrough not working with kapanga softphone
and eutelia provider
Description: 
If kapanga has direct access to the provider (no asterisk) I can send faxes
flawlessly, otherwise I get errors:

    -- Registered SIP '159' at 192.168.1.44 port 5060
       > Saved useragent "Kapanga Softphone Desktop Windows
1.00/2178d+1256057634_0022FA19F312_001377B41BFA" for peer 159
[Oct 21 00:53:58] NOTICE[12729]: chan_sip.c:17790
handle_response_peerpoke: Peer '159' is now Reachable. (147ms / 2000ms)
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
    -- Executing [<exten>@phones-sip:1] Dial("SIP/159-09614f80",
"SIP/eutelia/<exten>,60") in new stack
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
    -- Called eutelia/<exten>
<< [ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) ]
[SIP/eutelia-09610a90]
    -- SIP/eutelia-09610a90 is making progress passing it to
SIP/159-09614f80
[Oct 21 00:54:04] NOTICE[12753]: rtp.c:1130 process_rfc3389: Comfort noise
support incomplete in Asterisk (RFC 3389). Please turn off on client if
possible. Client IP: 192.168.1.44
<< [ TYPE: Unknown Frametype '10' (10) SUBCLASS: Unknown Subclass (114) ]
[SIP/159-09614f80]
<< [ TYPE: Unknown Frametype '10' (10) SUBCLASS: Unknown Subclass (114) ]
[SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/eutelia-09610a90]
<< [ TYPE: Unknown Frametype '10' (10) SUBCLASS: Unknown Subclass (114) ]
[SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/159-09614f80]
<< [ TYPE: Unknown Frametype '10' (10) SUBCLASS: Unknown Subclass (114) ]
[SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/eutelia-09610a90]
<< [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/eutelia-09610a90]
    -- SIP/eutelia-09610a90 answered SIP/159-09614f80
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/eutelia-09610a90]
<< [ TYPE: Unknown Frametype '10' (10) SUBCLASS: Unknown Subclass (114) ]
[SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/159-09614f80]
<< [ TYPE: Unknown Frametype '10' (10) SUBCLASS: Unknown Subclass (114) ]
[SIP/159-09614f80]
<< [ TYPE: Control (4) SUBCLASS: T38_Parameters/Negotiation Requested (24)
] [SIP/eutelia-09610a90]
<< [ TYPE: Control (4) SUBCLASS: Unknown control '20' (20) ]
[SIP/eutelia-09610a90]
<< [ TYPE: Control (4) SUBCLASS: T38_Parameters/Negotiated (24) ]
[SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/159-09614f80]
<< [ TYPE: Control (4) SUBCLASS: T38_Parameters/Negotiated (24) ]
[SIP/eutelia-09610a90]
<< [ TYPE: Unknown Frametype '10' (10) SUBCLASS: Unknown Subclass (114) ]
[SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/159-09614f80]
<< [ TYPE: Control (4) SUBCLASS: T38_Parameters/Terminated (24) ]
[SIP/eutelia-09610a90]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/159-09614f80]
<< [ TYPE: Unknown Frametype '10' (10) SUBCLASS: Unknown Subclass (114) ]
[SIP/159-09614f80]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/159-09614f80]
[Oct 21 00:54:35] WARNING[12729]: chan_sip.c:3557 retrans_pkt: Maximum
retries exceeded on transmission
781afd3767ab81d160f2ef2c6fd30037 at 217.133.76.61 for seqno 101 (Critical
Response) -- See doc/sip-retransmit.txt.
[Oct 21 00:54:35] WARNING[12729]: chan_sip.c:3584 retrans_pkt: Hanging up
call 781afd3767ab81d160f2ef2c6fd30037 at 217.133.76.61 - no reply to our
critical packet (see doc/sip-retransmit.txt).
  == Spawn extension (phones-sip, <exten>, 1) exited non-zero on
'SIP/159-09614f80'



This is my sip.conf:

[general]
t38pt_udptl=yes
language=it
context=default
bindport=5060
externhost=<host>
externrefresh=5
localnet=192.168.1.0/255.255.255.0
nat=yes
register=<number>:<password>@voip.eutelia.it:5060/<number>

[159]
language=it
type=friend
bindport=5060
context=phones-sip
host=dynamic
secret=<password>
nat=no
t38pt_udptl=yes
qualify=yes

[eutelia]
language=it
type=friend
host=voip.eutelia.it
username=<number>
fromuser=<number>
secret=<password>
port=5060
nat=yes
qualify=yes
canreinvite=no ;asterisk is always used as media proxy, I tried also with
yes
dtmfmode=rfc2833
insecure=invite,port
context=eutelia-in
t38pt_udptl=yes
====================================================================== 

---------------------------------------------------------------------- 
 (0112632) darkbasic (reporter) - 2009-10-22 16:11
 https://issues.asterisk.org/view.php?id=16107#c112632 
---------------------------------------------------------------------- 
I added sipdebug = yes, recordhistory = yes, dumphistory = yes to sip.conf
and I typed core set debug 4, core set verbose 4, sip set debug on, sip set
history on in asterisk CLI.
I attached /var/log/messages and /var/log/debug.
I don't think it's a provider issue because if I don't use kapanga as an
asterisk extentions it works flawlessly. Moreover a friend uses the same
provider with callweaver and it works fine.

Thank you,
Darkbasic 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-10-22 16:11 darkbasic      Note Added: 0112632                          
======================================================================




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