[Asterisk-Users] SIP vs. SIP :-(

[Igor Barsanti]

Resolved...

canreinvite=no

(i've put careinvite.... :-))

igor

On Tue, 2004-06-01 at 19:24, Igor Barsanti wrote:
> I'v a sip client and a sip trunk to FWD:
> 
> [general]
> port=5060
> context=default
> tos=reliability
> disallow=all
> allow=ulaw
> careinvite=no
> 
> [freeworlddialup]
> context=default
> type=friend
> username=MYUSERNAME
> secret=MYPASSWORD
> host=fwd.pulver.com
> 
> [igor]
> type=friend
> callerid="Me"
> host=dynamic
> dtmfmode=rfc2833
> careinvite=no
> 
> When i try to call a FWD number from SIP client i obtain a lot of
> build_route: messages from asterisk then the sip client die
> 
> .......
> Stopping retransmission on
> '569a2a5a77939bca491565ec50d0d3e5 at 82.51.138.189' of Request 104: Found
> build_route: Record-Route hop:
> <sip:421171 at 192.246.69.223;ftag=as61269cb9;lr=on>
> build_route: Contact hop: <sip:421171 at 65.39.205.121>
> ..............
> 
> ...with H.323 client all works perfectly. What's the problem ???
> 
> Igor
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