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Summary:ASTERISK-11262: One way audio problem
Reporter:Ravichandran Rajagopal (sunmoonstar)Labels:
Date Opened:2008-01-18 16:20:33.000-0600Date Closed:2011-06-07 14:03:23
Priority:MajorRegression?No
Status:Closed/CompleteComponents:General
Versions:Frequency of
Occurrence
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Issues:
Environment:Attachments:
Description:Hi,

I am getting an inbound call from a SIP trunk into my asterisk switch. When this call comes into a specific extension we play back an IVR. The user is asked to press a number on the key pad, when the user presses that number, the asterisk dials out a number through the SIP trunk. The issue is we are facing a one way audio. I have included the sip.conf and the extensions.conf in the additional information below for you to take a look at.
Thx
ravi

****** ADDITIONAL INFORMATION ******

Extensions.conf
[general]
static=yes
writeprotect=yes
clearglobalvars=no

[agnosco_mainmenu]
include => local-extensions
exten => s,1,Answer
exten => s,n,Background(agnosco_intro)
exten => s,n,WaitExten

;Dial said extensions
exten => 2,1,Dial(SIP/4026803515@pinpoint,30)
exten => 3,1,Dial(SIP/4022030123@pinpoint,30)
exten => 4,1,Dial(SIP/3086956544@pinpoint,30)
exten => 5,1,Dial(SIP/4028805362@pinpoint,30)

[pinpoint]
include => local-extensions
exten => 4025901000,1,Goto(1000,1)
exten => 4025901001,1,Goto(1001,1)
exten => 4025901002,1,Goto(1002,1)
exten => 4025901003,1,Goto(1003,1)
exten => 4025901008,1,Goto(1008,1)

exten => h,1,Hangup
exten => i,1,AbsoluteTimeout(15)
exten => i,2,Congestion
exten => i,3,Hangup
;

[agnosco]
include => local-extensions
exten => h,1,Hangup
exten => i,1,Congestion
exten => i,2,Hangup

exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)

; Record voice file to /tmp directory
exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/asterisk-recording:wav)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording)
exten => 205,5,wait(2)
exten => 205,6,Hangup

[local-extensions]
include => agnosco_mainmenu
exten => 1000,1,Goto(agnosco_mainmenu,s,1)

[default]
include => agnosco
sip.conf

[general]
context=pinpoint                ; Default context for incoming calls
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
allow=ulaw                      ; Allow codecs in order of preference
allow=alaw                      ; Allow codecs in order of preference

[pinpoint]
canreinvite=no
host=69.2.0.136

type=peer
user=phone

[agnosco]
canreinvite=no

context=from-trunk
dtmf=inband
dtmfmode=rfc2833
fromdomain=69.2.0.136
host=69.2.0.136
insecure=very
type=user
user=phone

[conference_bridge]
canreinvite=no
context=from-trunk
dtmf=inband
dtmfmode=rfc2833
fromdomain=69.2.0.136
host=69.2.0.136
insecure=very
type=user
user=phone
Comments:By: Joshua C. Colp (jcolp) 2008-01-18 16:33:23.000-0600

The issue tracker is not a support forum. Please seek help on the -users mailing list or #asterisk IRC channel with one way audio. It is quite a common issue.