Bug#462410: asterisk: astrerisk fails after update on codec conversion and crashes

Thu Jan 24 18:07:14 UTC 2008

Package: asterisk
Version: 1:1.4.17~dfsg-2
Severity: critical
Justification: breaks the whole system

After updating to 1.4.17 the system is crashing everytime it receives or
send a connection to the chan-capi. The Capi is connected to a Eicon
Diva Server 4BRI. This was working before. A call transfer from SIP to
SIP or IAX or voicemail does work. Transfer from Capi to IAX does lead
to a crash too.
In the chan capi alaw ist used, in sip alaw and ulaw.
I wonder about the messages (see below)

No translator path exists for channel type  SIP (native 65535) to 0
No translator path exists for channel type CAPI (native 8) to 0
Unable to find a codec translation path from unknown to gsm

Why is doesnt find any translator, as it worked before and another
question why it is crashing at the end.

here what happens:
 == ISDN1#02: Incoming call '04051452102' -> '519301'
     -- ISDN1#02: Updated channel name: CAPI/ISDN1#02/519301-1
     -- Executing [519301 at isdn-inbound:1] NoOp("CAPI/ISDN1#02/519301-1",
     "Anruf von 04051452102 fuer 519301") in new stack
     -- Executing [519301 at isdn-inbound:2]
     GotoIf("CAPI/ISDN1#02/519301-1", "0?cfanlage:zweitregel") in new
     stack
     -- Goto (isdn-inbound,519301,3)
     -- Executing [519301 at isdn-inbound:3]
     GotoIf("CAPI/ISDN1#02/519301-1", "0?cfnormal:normal") in new stack
     -- Goto (isdn-inbound,519301,4)
     -- Executing [519301 at isdn-inbound:4] Dial("CAPI/ISDN1#02/519301-1",
     "SIP/00|20|tT") in new stack
[Jan 24 18:57:55] WARNING[18385]: channel.c:3284
ast_request_with_uniqueid: No translator path exists for channel type
SIP (native 65535) to 0
[Jan 24 18:57:55] WARNING[18385]: app_dial.c:1210 dial_exec_full: Unable
to create channel of type 'SIP' (cause 58 - Bearer capability not
available)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [519301 at isdn-inbound:5] Dial("CAPI/ISDN1#02/519301-1",
"CAPI/ISDN2/01752768123/bol|20|r") in new stack
[Jan 24 18:57:55] WARNING[18385]: channel.c:3284
ast_request_with_uniqueid: No translator path exists for channel type
CAPI (native 8) to 0
[Jan 24 18:57:55] WARNING[18385]: app_dial.c:1210 dial_exec_full: Unable
to create channel of type 'CAPI' (cause 58 - Bearer capability not
available)
  == Everyone is busy/congested at this time (1:0/0/1)
  -- Executing [519301 at isdn-inbound:6]
  VoiceMail("CAPI/ISDN1#02/519301-1", "00|u") in new stack
  == ISDN1#02: Answering for 519301
[Jan 24 18:57:55] WARNING[18385]: channel.c:3059 set_format: Unable to
  find a codec translation path from unknown to gsm
[Jan 24 18:57:55] WARNING[18385]: file.c:871 ast_streamfile: Unable to
  open vm-theperson (format 0x0 (nothing)): No such file or directory
  == Spawn extension (isdn-inbound, 519301, 6) exited non-zero on
  'CAPI/ISDN1#02/519301-1'

Crash

-- System Information:
Debian Release: lenny/sid
  APT prefers testing
  APT policy: (990, 'testing')
Architecture: i386 (i686)

Kernel: Linux 2.6.18-4-686 (SMP w/1 CPU core)
Locale: LANG=de_DE.UTF-8, LC_CTYPE=de_DE.UTF-8 (charmap=UTF-8) (ignored: LC_ALL set to de_DE.utf8)
Shell: /bin/sh linked to /bin/bash

Versions of packages asterisk depends on:
ii  adduser               3.105              add and remove users and groups
ii  asterisk-config       1:1.4.17~dfsg-2    Configuration files for Asterisk
ii  asterisk-sounds-main  1:1.4.17~dfsg-2    Core Sound files for Asterisk (Eng
ii  libasound2            1.0.15-3           ALSA library
ii  libc-client2007       7:2007~dfsg-1      UW c-client library for mail proto
ii  libc6                 2.7-6              GNU C Library: Shared libraries
ii  libcap1               1:1.10-14          support for getting/setting POSIX.
ii  libct3                0.63-3.2           libraries for connecting to MS SQL
ii  libcurl3              7.17.1-1           Multi-protocol file transfer libra
ii  libgcc1               1:4.3-20080116-1   GCC support library
ii  libgsm1               1.0.12-1           Shared libraries for GSM speech co
ii  libiksemel3           1.2-3              C library for the Jabber IM platfo
ii  libkrb53              1.6.dfsg.3~beta1-2 MIT Kerberos runtime libraries
ii  libncurses5           5.6+20080105-1     Shared libraries for terminal hand
ii  libnewt0.52           0.52.2-11.1        Not Erik's Windowing Toolkit - tex
ii  libogg0               1.1.3-3            Ogg Bitstream Library
ii  libpopt0              1.10-3             lib for parsing cmdline parameters
ii  libpq5                8.2.6-1            PostgreSQL C client library
ii  libpri1.0             1.4.2-1            Primary Rate ISDN specification li
ii  libradiusclient-ng2   0.5.5-1            Enhanced RADIUS client library
ii  libsnmp15             5.4.1~dfsg-5       SNMP (Simple Network Management Pr
ii  libspeex1             1.1.12-3           The Speex Speech Codec
ii  libsqlite0            2.8.17-4           SQLite shared library
ii  libssl0.9.8           0.9.8g-3           SSL shared libraries
ii  libstdc++6            4.3-20080116-1     The GNU Standard C++ Library v3
ii  libtonezone1          1:1.4.7.1~dfsg-1   tonezone library (runtime)
ii  libvorbis0a           1.2.0.dfsg-3       The Vorbis General Audio Compressi
ii  libvorbisenc2         1.2.0.dfsg-3       The Vorbis General Audio Compressi
ii  libvpb0               4.2.23-1           Voicetronix telephony hardware use
ii  unixodbc              2.2.11-16          ODBC tools libraries
ii  zlib1g                1:1.2.3.3.dfsg-8   compression library - runtime

asterisk recommends no packages.

-- no debconf information