[ILUG] Asterisk woes

[Kevin Brennan]

SIP uses RTP/RTCP for the audio stream on separate random UDP ports.
If you connecting locally to the Asterisk server deactivate the 
firewall/iptables, if you are running Asterisk as a remotely hosted PBX 
it's likely your routers NAT is the issue - which is the case ?


On 06/12/2010 09:50, Niall O Broin wrote:
> I have been using Blueface as my phone provider for a few years now, with phone service at home provided through a Linksys WRTG54P2 which is a WiFi router with two integrated FXS ports, which is registering directly to Blueface. I want to start using an in-house asterisk server but I am having no success in getting the Linksys to work with it.
>
> I can setup calls, but I get no voice, in either direction. I can use soft phones registering to the asterisk server, both the Mac application Telephone and X-Lite, and they work perfectly, both for internal calls and for external calls, originating and receiving.
>
> I thought this might be some peculiarity of the Linksys so I replaced it with a Handytone HT502 2 port ATA adapter which I had handy ;-)  and sadly, it suffers from the exact same problem as the Linksys - call setup is fine, but no voice.  If I initiate a call between a hard phone and a soft phone tcpdump shows me UDB packets whizzing back and forth between the two, but I guess they're not speaking the same language, for whatever reason.
>
> The asterisk configuration as it differs from stock is below, with only one extension shown. Here's  sip.conf
>
> [general]
>
> register =>  BluefaceUser:BluefacePass at sip.blueface.ie/1000
> udpbindaddr=0.0.0.0
> allow=all
>
> [1000]
> type=friend
> context=phones
> host=dynamic
> secret=seekr1t
>
> [blueface]
> type=peer
> host=sip.blueface.ie
> username=BluefaceUser
> fromuser=BluefaceUser
> secret=BluefacePass
> insecure=invite
> context=incoming_calls
>
> and here's extensions.conf
>
> [globals]
>
> [general]
> autofallthrough=yes
>
> [default]
> exten =>  s,1,Verbose(1,Unrouted call handler)
> exten =>  s,n,Answer()
> exten =>  s,n,Wait(1)
> exten =>  s,n,Playback(tt-weasels)
> exten =>  s,n,Hangup()
>
> [incoming_calls]
> exten =>  1001,1,NoOp()
> exten =>  1001,n,Dial(SIP/1001,30)
> exten =>  _X.,1.NoOp()
> exten =>  _X.,n,Dial(SIP/1001)
>
> [outgoing_calls]
> exten =>  _X.,1,NoOp()
> exten =>  _X.,n,Dial(SIP/blueface/${EXTEN})
>
> [internal]
> exten =>  500,1,Verbose(1,Echo test application)
> exten =>  500,n,Echo()
> exten =>  500,n,Hangup()
>
> exten =>  1000,1,Verbose(1,Extension 1000)
> exten =>  1000,n,Dial(SIP/1000,30)
> exten =>  1000,n,Hangup()
>
> [phones]
> include =>  internal
> include =>  outgoing_calls
>
> Any suggestions as to where to look would be greatly appreciated.
>
>
>
> Niall
>
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