I am new to both Asterisk and Ubuntu (12.04.2 LTS server on a NAT LAN), so it has been quite easy for me to get lost trying to set my PBX even while following the “…Definitive Guide”.
After several (three to be exact) days of exhaustive online search, I am unable to find a solution to why there is no audio between my two Zoiper Communicator SIP softphones (one on a laptop, the other on a desktop; both running windows 7). Audio is fine between two IAX accounts and from IAX to SIP (and vice versa), on the same two softphones.
I am so far following the book almost verbatim, so the code is pretty basic at this point. I will truly appreciate your help resolving this issue.
My Hdw and software:
laptop: HP G72 notebook pc running windows 7 home premium
and Zoiper Communicator 2.05.11136 softphone.
cpu 3.06GHz, Memory 6GB, 64-bit operating system
Desktop: Micro Electronics PowerSpec running windows 7 professional
and Zoiper Communicator 2.05.11136 softphone.
cpu 2.30GHz, Memory 3GB, 64-bit operating system
PBX Server: Ubuntu 12.04.2 LTS (ONLY) on a
Gateway desktop: cpu 1GHz
I am running the latest Asterisk PBX which I believe is 11.5.
I am operating from a home network behind a NAT router (linksys Wrt54G.
I am not sure what file extension is allowed. Since text files are not allowed I had to cut and paste my configuration files. Please let me know if you need additional information.
[code]; EXTENSIONS.CONF
;============================================================
;
[globals]
LAPTOP=SIP/3000&IAX2/branch-office
BSMT=SIP/3001&IAX2/head-office
DAYO=SIP/3002&IAX2/dayo-iphone-4s
[LocalSets]
exten => 3000,1,Dial(${LAPTOP})
exten => laptop,1,Dial(${LAPTOP})
exten => 3001,1,Dial(${BSMT})
exten => bsmt,1,Dial(${BSMT})
exten => 3002,1,Dial(${DAYO})
exten => dayo,1,Dial(${DAYO})
exten => 4000,1,Dial(IAX2/head-office)
exten => 4001,1,Dial(IAX2/branch-office)
exten => 4002,1,Dial(IAX2/dayo-iphone-4s)
exten => 200,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
exten => 201,1,Goto(TestMenu,start,1) ; access the TestMenu context
exten => 4100,1,Dial(IAX2/head-office/4000)
exten => 4101,1,Dial(IAX2/branch-office/4001)
exten => 4101,1,Dial(IAX2/dayo-iphone-4s/4002)
exten => _4XXX,1,Dial(IAX2/head-office/${EXTEN})
[TestMenu]
exten => start,1,Answer()
same => n,Background(enter-ext-of-person)
same => n,WaitExten()
exten => 1,1,Dial(DAHDI/1,10)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => 2,1,Dial(SIP/Jane,10)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => 3,1,Dial(SIP/3001,10)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => i,1,Playback(pbx-invalid)
same => n,Goto(TestMenu,start,1)
exten => t,1,Playback(vm-goodbye)
same => n,Hangup()
[/code]
; sip.conf
; ---------------------------------------------------------------------------------------
[general]
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no
[office-phone](!)
description=office-phone
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw
allow=alaw
[3000](office-phone)
description=Akins Laptop
secret=xxxxxx
[3001](office-phone)
description=powerspec
secret=xxxxxx
[3002](office-phone)
description=Dayo's IPhone 4S
secret=xxxxxxHope I am in the right forum.