<div>you must add option insecure=very|yes|no in sip.conf, see <a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf">http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf</a> for more info
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<div>by default incoming calls goes into default context</div>
<div>have you checked if registration has occured in sipproxy?<br>check debug messages in asterisk console<br> </div>
<div><span class="gmail_quote">2006/8/9, kjcsb <<a href="mailto:kjcsb@orcon.net.nz">kjcsb@orcon.net.nz</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">I need to handle the following scenarios:<br>1. UA1 --> SIP Proxy --> Asterisk<br><br>2. UA2 --> SIP Proxy --> Asterisk --> PSTN gateway (SIP)
<br><br>I have configured a trunk to register with the SIP proxy:<br>trunk1<br>register=user1:password1@SIP.Proxy/DID1<br><br>UA1 calls <a href="mailto:user1@SIP.Proxy">user1@SIP.Proxy</a> and the call is recognised as being to DID1. I set
<br>up an inbound route for DID1 and route the call as appropriate. That deals<br>with scenario 1.<br><br>I then tried to configure another trunk to handle scenario 2:<br>trunk2<br>context=from-internal<br>host=SIP.Proxy<br>
type=peer<br>register=<a href="mailto:user2:password2@SIP.Proxy">user2:password2@SIP.Proxy</a><br><br>A call to PSTN1 from the UA is passed to the SIP proxy which recognises it<br>as PSTN call. The SIP proxy updates the From details and passes the call to
<br>Asterisk which (I presume) puts the call into the from-internal context and<br>dials the outbound route appropriately.<br><br>However that setup messes up scenario 1 which now gives a 404 back to UA1. I<br>presume Asterisk is not differentiating between a call made to user1 from
<br>UA1 and a call made to PSTN1 from user2. It's just seeing a call from<br>SIP.Proxy and putting it into the from-internal context.<br><br>Could anyone advise how I would set up Asterisk to cope with both these<br>scenarios? I could setup DID2 but I don't know how to pass the call onto the
<br>PSTN gateway. I am using AMP/FreePBX but if someone could advise the general<br>principles I would appreciate it.<br><br>Thanks<br><br>Cameron<br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by
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</a><br></blockquote></div><br>