Bug#557888: Playback() on an iax channel causes asterisk to hang indefinitly
Christoph Kling
christoph at familiekling.de
Wed Nov 25 02:49:04 UTC 2009
Package: asterisk
Version: 1:1.6.2.0~dfsg~rc1-1
Severity: important
Hello,
when I am using Playback() on an iax channel, the sound is very choppy at the
beginning. After some seconds, the playback stops and the playing asterisk
server does not continue to send audio to the client asterisk.
Here is the console log from the playing asterisk:
-- Executing [**20 at sip-in:1] Answer("IAX2/interlink-5309", "") in new stack
-- Executing [**20 at sip-in:2] Wait("IAX2/interlink-5309", "2") in new stack
-- Executing [**20 at sip-in:3] Playback("IAX2/interlink-5309", "agent-pass") in new stack
-- <IAX2/interlink-5309> Playing 'agent-pass.gsm' (language 'de')
Even after more than two minutes, there is neither any sound nor does the
playing asterisk server cancel in any way. My setup consists of two asterisk
servers being connected. The client asterisk connects the iax channel to a sip
channel. Enabling or disabling jitterbuffer does not change the behaviour.
tcpdump showed me that the client asterisk server is continuing to send packets
while the playing asterisk server does send some packets every five seconds or so.
I used gsm on both connections (asterisk <-> asterisk <-> sip) because the original
sound file is also gsm.
Tests with a direct iax connection from my sip client to the playing asterisk gave
the same result (client used: kiax/win32). If I use sip to connect to the playing
asterisk server directly, everything is fine! The problem occurs only with iax.
core show channel IAX2/interlink-5309 gives the following output (while hanging):
delta*CLI> core show channel iax2/interlink-5309
-- General --
Name: IAX2/interlink-5309
Type: IAX2
UniqueID: 1259116337.50
Caller ID: phi
Caller ID Name: Christoph Kling
DNID Digits: (N/A)
Language: de
State: Up (6)
Rings: 0
NativeFormats: 0x2 (gsm)
WriteFormat: 0x2 (gsm)
ReadFormat: 0x2 (gsm)
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: -1
Frames in: 1773
Frames out: 73
Time to Hangup: 0
Elapsed Time: 0h0m35s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: sip-in
Extension: **20
Priority: 3
Call Group: 0
Pickup Group: 0
Application: Playback
Data: agent-pass
Blocking in: ast_waitfor_nandfds
CDR Variables:
level 1: clid="xxx" <phi>
level 1: src=phi
level 1: dst=**20
level 1: dcontext=sip-in
level 1: channel=IAX2/interlink-5309
level 1: lastapp=Playback
level 1: lastdata=agent-pass
level 1: start=2009-11-25 03:32:17
level 1: answer=2009-11-25 03:32:17
level 1: duration=35
level 1: billsec=35
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1259116337.50
Please request further information from me if you need.
Regards,
Christoph Kling
-- System Information:
Debian Release: squeeze/sid
APT prefers testing
APT policy: (500, 'testing')
Architecture: i386 (i686)
Kernel: Linux 2.6.30.1
Locale: LANG=en_US, LC_CTYPE=en_US (charmap=ISO-8859-1)
Shell: /bin/sh linked to /bin/dash
Versions of packages asterisk depends on:
ii adduser 3.111 add and remove users and groups
ii asterisk-config 1:1.6.2.0~dfsg~rc1-1 Configuration files for Asterisk
ii asterisk-sounds-mai 1:1.6.2.0~dfsg~rc1-1 Core Sound files for Asterisk (Eng
ii dahdi 1:2.2.0-1 utilities for using the DAHDI kern
ii libasound2 1.0.21a-1 shared library for ALSA applicatio
ii libc-client2007b 8:2007b~dfsg-1.1+b1 c-client library for mail protocol
ii libc6 2.10.1-7 GNU C Library: Shared libraries
ii libcap2 1:2.17-2 support for getting/setting POSIX.
ii libcurl3 7.19.7-1 Multi-protocol file transfer libra
ii libgcc1 1:4.4.1-4 GCC support library
ii libglib2.0-0 2.22.2-2 The GLib library of C routines
ii libgmime-2.0-2a 2.2.22-4 MIME library
ii libgsm1 1.0.13-3 Shared libraries for GSM speech co
ii libiksemel3 1.2-4 C library for the Jabber IM platfo
ii libldap-2.4-2 2.4.17-2.1 OpenLDAP libraries
ii liblua5.1-0 5.1.4-5 Simple, extensible, embeddable pro
ii libncurses5 5.7+20090803-2 shared libraries for terminal hand
ii libnewt0.52 0.52.10-4.1 Not Erik's Windowing Toolkit - tex
ii libogg0 1.1.4~dfsg-1 Ogg bitstream library
ii libopenais2 0.83-1 Standards-based cluster framework
ii libpopt0 1.15-1 lib for parsing cmdline parameters
ii libpq5 8.4.1-1 PostgreSQL C client library
ii libpri1.4 1.4.10.2-1 Primary Rate ISDN specification li
ii libradiusclient-ng2 0.5.6-1 Enhanced RADIUS client library
ii libsdl1.2debian 1.2.13-5 Simple DirectMedia Layer
ii libsnmp15 5.4.1~dfsg-12 SNMP (Simple Network Management Pr
ii libspandsp1 0.0.5~pre4-1 Telephony signal processing librar
ii libspeex1 1.2~rc1-1 The Speex codec runtime library
ii libspeexdsp1 1.2~rc1-1 The Speex extended runtime library
ii libsqlite0 2.8.17-6 SQLite shared library
ii libss7-1 1.0.2-1 Signalling System 7 (ss7) library
ii libssl0.9.8 0.9.8k-5 SSL shared libraries
ii libstdc++6 4.4.1-4 The GNU Standard C++ Library v3
ii libsybdb5 0.82-6 libraries for connecting to MS SQL
ii libtiff4 3.9.2-1 Tag Image File Format (TIFF) libra
ii libtonezone2.0 1:2.2.0-1 tonezone library (runtime)
ii libvorbis0a 1.2.3-3 The Vorbis General Audio Compressi
ii libvorbisenc2 1.2.3-3 The Vorbis General Audio Compressi
ii libvpb0 4.2.42-1 Voicetronix telephony hardware use
ii libx11-6 2:1.2.2-1 X11 client-side library
ii libxml2 2.7.6.dfsg-1 GNOME XML library
ii unixodbc 2.2.11-21 ODBC tools libraries
ii zlib1g 1:1.2.3.3.dfsg-15 compression library - runtime
asterisk recommends no packages.
Versions of packages asterisk suggests:
ii asterisk-dev 1:1.6.2.0~dfsg~rc1-1 Development files for Asterisk
pn asterisk-doc <none> (no description available)
pn asterisk-h323 <none> (no description available)
pn ekiga <none> (no description available)
pn kphone <none> (no description available)
pn ohphone <none> (no description available)
pn twinkle <none> (no description available)
-- no debconf information
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