Summary: | ASTERISK-11262: One way audio problem | ||
Reporter: | Ravichandran Rajagopal (sunmoonstar) | Labels: | |
Date Opened: | 2008-01-18 16:20:33.000-0600 | Date Closed: | 2011-06-07 14:03:23 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | General |
Versions: | Frequency of Occurrence | ||
Related 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. |