[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 11:00 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

====================================================================== 

---------------------------------------------------------------------- 
 (0132697) jacobli (reporter) - 2011-03-07 11:00
 https://issues.asterisk.org/view.php?id=18242#c132697 
---------------------------------------------------------------------- 
Sorry. The patch has to be applied to channels:
--- channels/chan_dahdi.c
+++ channels/chan_dahdi.c
@@ -9056,6 +9056,8 @@
         return 0;
     }
     if ((frame->subclass.codec != AST_FORMAT_SLINEAR) &&
+ (frame->subclass.codec != AST_FORMAT_ILBC) &&
+ (frame->subclass.codec != AST_FORMAT_GSM) &&
         (frame->subclass.codec != AST_FORMAT_ULAW) &&
         (frame->subclass.codec != AST_FORMAT_ALAW)) {
         ast_log(LOG_WARNING, "Cannot handle frames in %s format\n",
ast_getformatname(frame->subclass.codec));

And the issue is not related to DISA only.
And probably there are something behind the patch what the patch is
masking but I'm running several asterisk the patch was applied to. Didn't
run into any problem. Hope developers will investigate the issue and close
it. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-03-07 11:00 jacobli        Note Added: 0132697                          
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