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

[George Pajari]

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