[asterisk-users] Handling inbound and outbound calls passed from a proxy

Fran Oliveira tech.oliveira at gmail.com
Wed Aug 9 05:39:51 MST 2006


you must add option insecure=very|yes|no in sip.conf, see
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf for
more info
by default incoming calls goes into default context
have you checked if registration has occured in sipproxy?
check debug messages in asterisk console

2006/8/9, kjcsb <kjcsb at orcon.net.nz>:
>
> I need to handle the following scenarios:
> 1. UA1 --> SIP Proxy --> Asterisk
>
> 2. UA2 --> SIP Proxy --> Asterisk --> PSTN gateway (SIP)
>
> I have configured a trunk to register with the SIP proxy:
> trunk1
> register=user1:password1 at SIP.Proxy/DID1
>
> UA1 calls user1 at SIP.Proxy and the call is recognised as being to DID1. I
> set
> up an inbound route for DID1 and route the call as appropriate. That deals
> with scenario 1.
>
> I then tried to configure another trunk to handle scenario 2:
> trunk2
> context=from-internal
> host=SIP.Proxy
> type=peer
> register=user2:password2 at SIP.Proxy
>
> A call to PSTN1 from the UA is passed to the SIP proxy which recognises it
> as PSTN call. The SIP proxy updates the From details and passes the call
> to
> Asterisk which (I presume) puts the call into the from-internal context
> and
> dials the outbound route appropriately.
>
> However that setup messes up scenario 1 which now gives a 404 back to UA1.
> I
> presume Asterisk is not differentiating between a call made to user1 from
> UA1 and a call made to PSTN1 from user2. It's just seeing a call from
> SIP.Proxy and putting it into the from-internal context.
>
> Could anyone advise how I would set up Asterisk to cope with both these
> scenarios? I could setup DID2 but I don't know how to pass the call onto
> the
> PSTN gateway. I am using AMP/FreePBX but if someone could advise the
> general
> principles I would appreciate it.
>
> Thanks
>
> Cameron
>
>
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