[Asterisk-Users] internal & external SIP

[Soren Rathje]

From: "Jon Lawrence"

> 
> Okay, I've made some changes. I've moved the local phones to public IP's.
> So now everything is connecting effectively from the internet to the * box.
> Things are still the same as before - I can initiate calls from local phones 
> to remote ones.
> If a remote phone tries to initiate the call, the internal phone rings. When I 
> pickup the internal phone, the call isn't completed.
> 
.. snip ..
>
>  to 82.145.37.29:5060
> Jul  9 12:41:49 WARNING[5126]: chan_sip.c:495 retrans_pkt: Maximum retries 
> exceeded on call b736bef35cf69297 at 82.145.37.29 for seqno 7712 (Response)
> set_destination: Parsing <sip:2000 at 81.168.4.69> for address/port to send to
> set_destination: set destination to 81.168.4.69, port 5060
> Reliably Transmitting:
> BYE sip:2000 at 81.168.4.69 SIP/2.0
> Via: SIP/2.0/UDP 81.168.4.67:5060;branch=z9hG4bK201b0b71
> From: "2003" <sip:2003 at 81.168.4.67>;tag=as3f8ccbff
> To: <sip:2000 at 81.168.4.69>;tag=0939785f3bc7641e
> Contact: <sip:2003 at 81.168.4.67>
> Call-ID: 666e652b583d10060af59dcb11feffee at 81.168.4.67
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Content-Length: 0
> 

What are your codec settings in sip.conf ??

Could you try (can be set at client level):

disallow=all
allow=ulaw

-- Soren