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

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Holger Wegner

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Jan 24, 2008, 1:20:14 PM1/24/08
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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@isdn-inbound:1] NoOp("CAPI/ISDN1#02/519301-1",
"Anruf von 04051452102 fuer 519301") in new stack
-- Executing [519301@isdn-inbound:2]
GotoIf("CAPI/ISDN1#02/519301-1", "0?cfanlage:zweitregel") in new
stack
-- Goto (isdn-inbound,519301,3)
-- Executing [519301@isdn-inbound:3]
GotoIf("CAPI/ISDN1#02/519301-1", "0?cfnormal:normal") in new stack
-- Goto (isdn-inbound,519301,4)
-- Executing [519301@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@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@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

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Holger Wegner

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Jan 26, 2008, 6:40:11 AM1/26/08
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Hi,
I build the asterisk-chan-capi package from new with:
- apt-get update

- apt-get build-dep asterisk-chan-capi

- apt-get source --build asterisk-chan-capi

and installed it. Now it works again for me.

Holger

Faidon Liambotis

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Jan 26, 2008, 5:00:22 PM1/26/08
to
Holger Wegner wrote:
> Hi,
> I build the asterisk-chan-capi package from new with:
> - apt-get update
>
> - apt-get build-dep asterisk-chan-capi
>
> - apt-get source --build asterisk-chan-capi
>
> and installed it. Now it works again for me.
Unfortunately, this means that at some point we (or probably upstream)
broke Asterisk's ABI.

You said that your setup broke when you upgaded to 1.4.17.
Which version did you have before?

Thanks,
Faidon

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