[asterisk-users] SIP trunks: order or type

[Marco Mouta]

Just a question:

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 <shaun.hofer at altcall.com> 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 '79119-3364165035-362070 at x.x.x.x' of Response 1: Match
> Found
>
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-- 
Com os melhores cumprimentos,

Marco Mouta
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