[Asterisk-Users] Why Won't Asterisk REINVITE?

[Julian J. M.]

Try removing the Answer() before the Dial... e.g.:

[spa2100]

exten => _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN})
exten => _X.,2,Dial(SIP/netvoice-102)
exten => _X.,3,Hangup

Regards
   Julian J. M.


On 12/9/05, George Pajari <George.Pajari at netvoice.ca> wrote:
> Eric "ManxPower" Wieling wrote:
>
> > T/t/H/h and other options to Dial require Asterisk to stay in the RTP
> > stream.
>
> Understood but already checked as not being the cause. Thanks for the
> suggestion, though.
>
> Here is our entire extensions.conf context:
>
> [spa2100]
>
> exten => _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN})
> exten => _X.,2,Answer
> exten => _X.,3,Wait(2)
> exten => _X.,4,Dial(SIP/netvoice-102)
> exten => _X.,5,Hangup
>
> where
>
> [netvoice-102]
> accountcode=netvoice-102
> callerid=NETVOICE COMMS <604 484 8647>
> username=netvoice-102
> type=friend
> host=dynamic
> dtmfmode=rfc2833
> nat=no
> qualify=no
> mailbox=102
> context = netvoice-internal
> canreinvite=yes
> disallow=all
> allow=ulaw
>
> Here is a "sip show channels" during a call:
>
> aa.bb.cc.39    netvoice-1  7f6a484c36f  00103/00000   ulaw
> aa.bb.cc.40    nvc.test.a  6cfe5077-2f  00103/00102   ulaw
>
> --
> George Pajari, netVOICE communications    604 484 VOIP (484 8647 x102)
> Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
>                   www.netvoice.ca  www.ip-centrex.ca
>       www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
>
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