Don't you need type=user to receive in this trunk?
As far as I know, peer is where you dial calls, and user is where calls can be placed.
To outbound a call from you * box via SIP trunk, this trunk must be type=peer or type=friend
To inbound calls to * box via SIP trunk , this trunk must be type=user or type=friend.
"friend=user+peer"
peers cannot place calls into the Asterisk server
http://www.asteriskpbx.com/
Best regards,
Marco Mouta
On 8/10/06, Shaun Hofer <[EMAIL PROTECTED]
> wrote:
I have two trunks to the same machine (x.x.x.2), one is type=friend, other is
type=peer. Asterisk seems to choose which trunk to use by the order by which
they are set out in sip.conf.
When a incoming call comes into Asterisk, it always uses the last trunk. My
understanding was that a peer trunk can't receive incoming calls. Does
Asterisk ignore the type when dealing with incoming calls from the same
host/machine ?
I want all incoming calls to use the back-trunk only. When I change the order
around from what it looks like below it works perfectly. I've been told that
order of things appearing in sip.conf should not matter.
--Shaun
sip.conf:
[back-trunk]
type = friend
username = 8880006111
secret = vvvvvv
host = x.x.x.2
dtmfmode = rfc2833
nat = no
canreinvite = no
insecure = port,invite
qualify = no
disallow = all
allow = ulaw
allow = alaw
allow = g729
context = shared-back-trunk-incoming
[back-trunk-ulaw]
type = peer
username = 8880006113
secret = vvvvvv
host = x.x.x.2
dtmfmode = rfc2833
nat = no
canreinvite = no
insecure = port,invite
qualify = no
disallow = all
allow = ulaw
context = shared-back-trunk-ulaw-incoming
Asterisk CLI:
Aug 10 12:17:15 DEBUG[21756]: chan_sip.c:7242 check_user_full: Setting NAT on
RTP to 0
Aug 10 12:17:15 DEBUG[21756]: chan_sip.c:10497 handle_request_invite: Checking
SIP call limits for device 8880006113
Aug 10 12:17:15 DEBUG[21756]: chan_sip.c:1401 __sip_ack: Stopping
retransmission on '[EMAIL PROTECTED]' of Response 1: Match
Found
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Com os melhores cumprimentos,
Marco Mouta
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