No audio via SIP but IAX works fine using Zoiper

Asterisk Asterisk General
1

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.

2

Interesting to hear you have problems, but without debugging information and/or detailed hardware and software configuration information, there is nothing we can do but sympathize.

This, of course, is the wrong forum if you are asking for support.

3

Thanks for the sympathy and sorry for the intrusion.

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=xxxxxx

Also, Could you direct me to the proper forum.

4

Did you check the firewall rules on the Asterisk server and both PC? For SIP you need to allow traffic on UDP ports 5060 and range of UDP ports 10 000 - 20 000.

Are the PC’s and server part of the same LAN or is the server on a public IP address?