[asterisk-bugs] [Asterisk 0018242]: DISA "Cannot handle frames in gsm format"
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Mar 6 20:52:26 CST 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18242
======================================================================
Reported By: gentlec
Assigned To:
======================================================================
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:
======================================================================
Date Submitted: 2010-11-02 12:53 CDT
Last Modified: 2011-03-06 20:52 CST
======================================================================
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
======================================================================
----------------------------------------------------------------------
(0132679) clint (reporter) - 2011-03-06 20:52
https://issues.asterisk.org/view.php?id=18242#c132679
----------------------------------------------------------------------
More information:
Issue occurs on any DIAL via DAHDI, not only in DISA.
Issue only occurs if Asterisk is generating indication.
eg. ringback (r flag to dial command) or music on hold (m flag to dial)
The issue will occur with ANY non ulaw codec over IAX. Tested with GSM,
G722, G729.
Tested in 1.8.3.
Requires CALLER to come in over IAX, and be routed to DAHDI channel.
Examples:
Failure, Asterisk generating indication via "r" flag to dial command:
-- Executing [151 at xenir-group:2] Dial("IAX2/banbury-279",
"Dahdi/G1/1XXXXXXXXXX,24,tr") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/1XXXXXXXXXX
[Mar 6 20:39:05] WARNING[2994]: chan_dahdi.c:9061 dahdi_write: Cannot
handle frames in g722 format
[Mar 6 20:39:05] WARNING[2994]: app_dial.c:1410 wait_for_answer: Unable
to forward voice or dtmf
-- Hungup 'DAHDI/i1/1XXXXXXXXXX-a'
== Everyone is busy/congested at this time (1:0/0/1)
Failure, music on hold ("m" flag to dial):
-- Executing [151 at xenir-group:2] Dial("IAX2/banbury-402",
"Dahdi/G1/1XXXXXXXXXX,24,tm") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/1XXXXXXXXXX
-- Started music on hold, class 'default', on IAX2/banbury-402
[Mar 6 20:37:43] WARNING[2987]: chan_dahdi.c:9061 dahdi_write: Cannot
handle frames in g722 format
[Mar 6 20:37:43] WARNING[2987]: app_dial.c:1410 wait_for_answer: Unable
to forward voice or dtmf
-- Hungup 'DAHDI/i1/1XXXXXXXXXX-9'
== Everyone is busy/congested at this time (1:0/0/1)
-- Stopped music on hold on IAX2/banbury-402
Success, passthrough indication (no "r" flag):
-- Executing [151 at xenir-group:2] Dial("IAX2/banbury-5514",
"Dahdi/G1/1XXXXXXXXXX,24,t") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/1XXXXXXXXXX
-- DAHDI/i1/1XXXXXXXXXX-8 is proceeding passing it to
IAX2/banbury-5514
-- IAX2/banbury-5514 requested special control 20, passing it to
DAHDI/i1/1XXXXXXXXXX-8
-- IAX2/banbury-5514 requested special control 20, passing it to
DAHDI/i1/1XXXXXXXXXX-8
-- DAHDI/i1/1XXXXXXXXXX-8 is making progress passing it to
IAX2/banbury-5514
This probably increases the severity of this bug a bit.
Issue History
Date Modified Username Field Change
======================================================================
2011-03-06 20:52 clint Note Added: 0132679
======================================================================
More information about the asterisk-bugs
mailing list