[ILUG] Asterisk woes

[Kevin Brennan]

It's very unlikely to be the SIP trunk you have to Blueface so IAX 
should not be required. Asterisk is acting like a B2BUA (Back to back 
user agent) so there's no reason why an ATA unit connected locally would 
not work and a soft phone would (to Blueface) unless you have a local 
issue. Keep your VoIP provider out of the picture and get local calls 
between extensions working correctly first (unless I have completely 
miss-read your email I understand calls between local extensions are not 
working for hardware).

Basically of you have a set up like this

Asterisk ---- NAT/Firwall ---- UA (ATA)

it can cause problems without some tweaks.

If you are running a hosted Asterisk with many remote extensions it can 
be a little trickier as there definitely will be NAT's and each will be 
a little different. If you post how your setup looks it will be easier 
to provide advice.

/KB.


On 06/12/2010 10:46, Barry O'Donovan wrote:
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> Niall,
>
> Most likely NAT issues with the RTP streams in SIP.
>
> Blueface very helpfully provide IAX trunks when requested. IAX packets
> bundle the messaging and RTP into the same packets avoiding NAT issues.
>
>   - Barry
>
> On 06/12/10 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
>>
>
> - --
>
>
> Kind regards,
> Barry O'Donovan
>
> http://www.opensolutions.ie/
>
> Mob: +353 86 801 7669
> Tel: +353  1 685 4220
> SIP: barry at opensolutions.ie
> Fax: +353  1 685 4263
>
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