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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