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Asterisk: The phone rings, and nothing is heard, silence on the phone, how to fix it?
test.call which I throw in /var/spoov/asterisk/outgoing:
Channel: SIP/sipnet/###########
Callerid: ##########
MaxRetries: 2
RetryTime: 20
WaitTime: 60
Context: default
Extension: s
Priority: 1
extensions.conf:[default]
exten => s,1,Answer()
exten => s,n,Playback(/home/krll/finish)
exten => s,n,Hangup()
Asterisk CLIkrll: ~ $ sudo asterisk -rvvvvv
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Asterisk 11.7.0~dfsg-1ubuntu1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.7.0~dfsg-1ubuntu1 currently running on krll (pid = 1441)
[Apr 23 19:28:18] WARNING[1480]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test
.call: Operation not permitted
-- Attempting call on SIP/sipnet/89991628348 for [email protected]:1 (Retry 1)
== Using SIP RTP CoS mark 5
> Channel SIP/sipnet-00000002 was answered
-- Executing [[email protected]:1] Answer("SIP/sipnet-00000002", "") in new stack
-- Executing [[email protected]:2] Playback("SIP/sipnet-00000002", "/home/krll/finish") in new stack
[Apr 23 19:28:31] WARNING[1513][C-00000002]: channel.c:5309 set_format: Unable to find a codec translation path from (
g729) to (alaw|slin)
[Apr 23 19:28:31] WARNING[1513][C-00000002]: file.c:1017 ast_streamfile: Unable to open /home/krll/finish (format (g72
9)): No such file or directory
[Apr 23 19:28:31] WARNING[1513][C-00000002]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/sipnet-000
00002 for /home/krll/finish
-- Executing [[email protected]:3] Hangup("SIP/sipnet-00000002", "") in new stack
== Spawn extension (default, s, 3) exited non-zero on 'SIP/sipnet-00000002'
[Apr 23 19:28:31] NOTICE[1513]: pbx_spool.c:402 attempt_thread: Call completed to SIP/sipnet/89991628348
krll*CLI>
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sipnet wants the g729 codec
What Asterisk says:
Unable to find a codec translation path from (g729) to (alaw|slin)
Put codec
cd /usr/lib64/asterisk/modules/ (just lib if you have x86)
asterisk -rvvvv
cli > !wget asterisk.hosting.lv/bin/codec_g729-ast110-gcc4-gli... (here you need to choose the right asterisk version, architecture, processor)
cli> module load codec_g729-ast110-gcc4-glibc-x86_64-core2-sse4 .so
cli> core show translation (to check installed codecs)
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