[asterisk-bugs] [Asterisk 0018242]: DISA "Cannot handle frames in gsm format"

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18242 
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Reported By:                gentlec
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18242
Category:                   Applications/app_disa
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.8.0 
JIRA:                       SWP-2592 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-11-02 12:53 CDT
Last Modified:              2011-03-07 11:58 CST
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Summary:                    DISA "Cannot handle frames in gsm format"
Description: 
When I call into my Asterisk box via a VoIP line (in this case it's
incoming IP from Vitelity using gsm codec) and then try to make an outgoing
DISA call over my PSTN I get the following:

[Nov  1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot
handle frames in gsm format
[Nov  1 15:12:54] WARNING[17694]: app_dial.c:1401 wait_for_answer: Unable
to forward voice or dtmf

It looks like asterisk is not converting the gsm frames to whatever it
needs to send over the PSTN.  I never had this problem with the 1.6.x
series but it started as soon as I upgraded to 1.8.0 and dahdi-2.4.0.  My
Asterisk machine has a TDM-410 card installed for the interface to the
PSTN.  Running on Ubuntu server 10.10.  DAHDI and Asterisk compiled from
source tarballs.  Please let me know if I didn't include something that you
need.

Here is the call log of the failed DISA call:

[Nov  2 12:12:58] VERBOSE[22688] pbx.c:     -- Executing
[XXXXXXXXXX at disa:16] Dial("SIP/xxxxxxx-00000000",
"dahdi/1/XXXXXXXXXX,60,r") in new stack
[Nov  2 12:12:58] DEBUG[22688] sig_analog.c: analog_available 1
[Nov  2 12:12:58] DEBUG[22688] sig_analog.c: analog_request 1
[Nov  2 12:12:58] DEBUG[22688] sig_analog.c: CALLING CID_NAME: Chris
Gentle CID_NUM:: XXXXXXXXXX
[Nov  2 12:12:58] DEBUG[22688] sig_analog.c: Dialing 'XXXXXXXXXX'
[Nov  2 12:12:58] VERBOSE[22688] app_dial.c:     -- Called 1/XXXXXXXXXX
[Nov  2 12:12:58] WARNING[22688] chan_dahdi.c: Cannot handle frames in gsm
format
[Nov  2 12:12:58] WARNING[22688] app_dial.c: Unable to forward voice or
dtmf
[Nov  2 12:12:58] DEBUG[22688] sig_analog.c: analog_hangup 1
[Nov  2 12:12:58] VERBOSE[22688] sig_analog.c:     -- Hanging up on
'DAHDI/1-1'
[Nov  2 12:12:58] VERBOSE[22688] chan_dahdi.c:     -- Hungup 'DAHDI/1-1'
[Nov  2 12:12:58] VERBOSE[22688] app_dial.c:   == Everyone is
busy/congested at this time (1:0/0/1)
[Nov  2 12:12:58] VERBOSE[22688] pbx.c:     -- Executing
[XXXXXXXXXX at disa:17] Hangup("SIP/xxxxxxx-00000000", "") in new stack
[Nov  2 12:12:58] VERBOSE[22688] pbx.c:   == Spawn extension (disa,
XXXXXXXXXX, 17) exited non-zero on 'SIP/xxxxxxx-00000000'

My sip profile for the incoming vitelity connection has the following
codec related settings:

disallow=all
allow=gsm

My chan_dahdi config looks like this:

[channels]
context=incoming
signalling=fxs_ks
toneduration=300
usecallerid=yes
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=no
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=4.0
txgain=1.0
group=1
callgroup=1
pickupgroup=1
immediate=no
adsi=yes
progzone=us
callerid=asreceived
channel => 1

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---------------------------------------------------------------------- 
 (0132701) gentlec (reporter) - 2011-03-07 11:58
 https://issues.asterisk.org/view.php?id=18242#c132701 
---------------------------------------------------------------------- 
I'll be happy to, if I can figure out how.  I can't seem to find a way to
edit the report.  Do I have permission to do that as the reporter?  What am
I missing? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-03-07 11:58 gentlec        Note Added: 0132701                          
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