[asterisk-bugs] [Asterisk 0013405]: [patch] T38 gateway
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Apr 20 05:18:33 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=13405
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Reported By: dafe_von_cetin
Assigned To:
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Project: Asterisk
Issue ID: 13405
Category: Applications/app_fax
Reproducibility: N/A
Severity: feature
Priority: normal
Status: confirmed
Asterisk Version: SVN
JIRA: SWP-115
Regression: No
Reviewboard Link: https://reviewboard.asterisk.org/r/459/
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 140548
Request Review:
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Date Submitted: 2008-08-30 16:44 CDT
Last Modified: 2010-04-20 05:18 CDT
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Summary: [patch] T38 gateway
Description:
Hi all,
I'm sending you patch containing new application app_faxgateway.c
("FaxGateway") which is able to mediate T30 to T38 and vice versa.
Feature is using spands library (I used spandsp-0.0.4pre18 and
spandsp-0.0.5pre4).
Best regards
Daniel.
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(0120628) tom_m (reporter) - 2010-04-20 05:18
https://issues.asterisk.org/view.php?id=13405#c120628
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Hi,
Can anybody please point out the best way to handle voice (features like
transfer etc) and t.38 gatewaying (PSTN<->SIP) in the dial plan?
When the call is placed, I do not know whether I am dialing for voice or
fax endpoints, so I need to dial with the b option in case t.38 bridging is
required.
exten => _X.,n,Dial(DAHDI/group1/${EXTEN},,b)
According to the documentation (and my experience) call features like
transfer are lost when I use the b option.
> b: Bridge T.38 Fax. This disables other call features that are usually
> available after the calling party answers. This includes transfer,
call
> and DTMF recording, etc.
The transfer of a voice call leads to a hangup.
I can reproduce this wit both libss7 versions, and different E1 hardware.
Am I missing something? How can I get around this? Application FaxGateway
cause same problem. I would like to prepare for t.38 bridging in case I
have a fax call on hand, but not lose voice call transfer capability (or
other voice features) if the call tuns out to be a voice call.
Here’s the associated debug output for a voice call that gets hungup on
attended transfer.
-- Executing [${CALLED_NUMBER}@dial:10]
Dial("SIP/sip_proxy1-000012ac", "Dahdi/group1/${CALLED_NUMBER},,b") in new
stack
-- Called group1/${CALLED_NUMBER}
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/120-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/120-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/120-1]
.
.
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/120-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/120-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/120-1]
<< [ TYPE: Control (4) SUBCLASS: Unknown control '15' (15) ]
[DAHDI/120-1]
-- DAHDI/120-1 is proceeding passing it to SIP/sip_proxy1-000012ac
<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [DAHDI/120-1]
-- DAHDI/120-1 is ringing
Unhandled optional parameter 0x2d 'Unknown'
[0x0 0x5a ]
Unhandled optional parameter 0x39 'Parameter Compatibility Information'
[0x2d 0x40 0x80 ]
<< [ TYPE: Control (4) SUBCLASS: Answer (4) ] [DAHDI/120-1]
-- DAHDI/120-1 answered SIP/sip_proxy1-000012ac
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/sip_proxy1-000012ac]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/sip_proxy1-000012ac]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/sip_proxy1-000012ac]
>> [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/120-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/sip_proxy1-000012ac]
>> [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/120-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/sip_proxy1-000012ac]
>> [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/12-1]
Unhandled optional parameter 0x2c 'Generic Notification Indication'
[0xf9 ]I>
Unhandled optional parameter 0x39 'Parameter Compatibility Information'
[0x2c 0x50 0x80 ]
<< [ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) ]
[DAHDI/120-1]
>> [ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) ]
[SIP/sip_proxy1-000012ac]
-- Hungup 'DAHDI/120-1'
== Spawn extension (com_hangup, h, 16) exited non-zero on
'SIP/sip_proxy1-000012ac'
Issue History
Date Modified Username Field Change
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2010-04-20 05:18 tom_m Note Added: 0120628
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