<div>Please give me some heads up.</div>
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<div>I'm having troube setting up my asterisk connecting to my SIP provider (SP). Here's the setup.</div>
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<div>in sip.conf</div>
<div>I register asterisk to SP using</div>
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<div>register => <a href="mailto:15551234567:pwd123@sip.abc.com">15551234567:pwd123@sip.abc.com</a></div>
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<div>everything was ok for incoming call until I want to dial out using the same line. since in order to dial out, i added in sip.conf</div>
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<div>[sip_out_to_SP]</div>
<div>type=peer<br>username=15551234567<br>fromuser=15551234567<br>secret=pwd123<br>host=<a href="http://sip.abc.com">sip.abc.com</a></div>
<div>port=5160 </div>
<div>usereqphone=yes <br>call-limit=5 </div>
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<div>Then I can make phone call out when routed to this channel SIP/sip_out_to_SP. But I no longer able to receive calls. The sniffer told me when my SP send asterisk an invite, asterisk returns a 407 to authenticate SP, which kills the connection.
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<div>I tried to change "type =friend" but doesn't work. I have to either remove "secret= " or "host=" to make my incoming call back.</div>
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<div>So currently, I can only either dial out or receive calls on this sip line. not both. (I'm not talking about at the same time). Can anyone give me a hint. REALLY APPRECIATE IT.</div>
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<div>Michael</div>