[Asterisk-Users] Unable to set audio mode on channel 1 + ring after hangup + moh +
echo
Frank Contrepois
f.contrepois at telvia.it
Wed Mar 26 07:01:51 MST 2003
Hi
I use a Wildcard X101P as FXO and a phone connected to an ATA186 using
SIP to connect to *.
phone ----- ata186 ----- * ----- X101P ----- pstn
On every call using the X101P I get
"Unable to set audio mode on channel 1" but I got no problem
Why sometimes after receiving a call the phone ring one time (only)???
This is the log of a standard call coming from outside
-- Starting simple switch on 'Zap/1-1'
-- Executing Macro("Zap/1-1", "stdUsr|211|SIP/211&SIP/213") in new stack
-- Executing Dial("Zap/1-1", "SIP/211&SIP/213|20|t|T") in new stack
-- Called 211
-- Called 213
-- SIP/211-d36b is ringing
-- SIP/213-89ec is ringing
-- SIP/211-d36b answered Zap/1-1
== Spawn extension (macro-stdUsr, s, 1) exited non-zero on 'Zap/1-1'
in macro 'stdUsr'
== Spawn extension (out2in, s, 1) exited non-zero on 'Zap/1-1'
WARNING[15371]: File chan_zap.c, Line 1876 (zt_setoption): Unable to set
audio mode on channel 1
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
WARNING[16395]: File chan_zap.c, Line 4041 (ss_thread): CallerID
returned with error on channel 'Zap/1-1'
-- Executing Macro("Zap/1-1", "stdUsr|211|SIP/211&SIP/213") in new stack
-- Executing Dial("Zap/1-1", "SIP/211&SIP/213|20|t|T") in new stack
-- Called 211
-- Called 213
-- SIP/211-c794 is ringing
-- SIP/213-35eb is ringing
== Spawn extension (macro-stdUsr, s, 1) exited non-zero on 'Zap/1-1'
in macro 'stdUsr'
== Spawn extension (out2in, s, 1) exited non-zero on 'Zap/1-1'
WARNING[16395]: File chan_zap.c, Line 1876 (zt_setoption): Unable to set
audio mode on channel 1
-- Hungup 'Zap/1-1'
What is the problem for the warning?
Third question, do I need a working soundcard to use music on hold?
I have a big echo on my side when talking using the fxo
(my zapata.conf)
;
; Zapata telephony interface
;
; Configuration file
[channels]
;of course we want to run the software echo canceller
echocancel=yes
echocancelwhenbridged=yes
language=en
context=out2in
signalling=fxs_ls
channel=1
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
;we can tune the gain on incoming and outgoing audio
;adjust these if you have a device that seems 'quiet' or 'loud'
rxgain=0.0
txgain=0.0
When compiling I use the
KFLAGS+=-DECHO_CAN_MARK2.
Thanks
Frank Contrepois
Coblan srl
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