Hey,<div> <div>I don;t think asterisk-guru could've been wrong on this one - possibly different scenario than your's. Anyway I see what you did there ! There is no need for separate context for incoming or outgoing if you don't want. What you are doing is <b>exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}<span class="Apple-style-span" style="background-color: rgb(255, 0, 0);">@1401</span></b><b>) </b></div>
<div><b><br></b></div><div>When you defined the SIp user/peer [1401] you stated context for handling dial request as "my-office" and when you tried dialling out you told asterisk to dial the requested number located at 1401 which should've been @<<a href="http://IP.OF.Grandstream.GW">IP.OF.Grandstream.GW</a>> if calls need to be dialed to gateway and If your gateway just accepts SIP based (w/o auth) calls. </div>
<div><br></div><div><b>exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}<span class="Apple-style-span" style="background-color: rgb(51, 255, 51);">@<a href="http://192.168.14.?">192.168.14.?</a>??</span></b><b>) </b></div><div><b><br>
</b></div><div>If your gateway shows attitude in serving direct request you may need to create user in gateway and telling asterisk to register on Grandstream as a user and dial-out using that user like.</div><div><br></div>
<div><b>exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}<span class="Apple-style-span" style="background-color: rgb(51, 255, 51); ">@gstream-user</span></b><b>) </b></div><div><b><br></b></div><div>There could be more possible alternatives to successfully dial-out using one context for handling incoming an out going/ preferred is you create separate contexts.</div>
<div><br></div><div>Regards,</div><div>- Sammy</div><div><br><div class="gmail_quote">On Tue, Sep 20, 2011 at 8:13 PM, Samuel Sappa <span dir="ltr"><<a href="mailto:cihuy916@gmail.com">cihuy916@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Sorry if this question already asked.<br>
I'm implementing Voip with asterisk and grandstream gxw4108, according<br>
from the manual, for connecting with PSTN I must configure one SIP<br>
account and assign that for dialing the PSTN so in my sip.conf I<br>
configure SIP account(extension) :<br>
<br>
[1401]<br>
type=friend<br>
username=1401<br>
secret=1401<br>
host=dynamic<br>
context=my-office<br>
insecure=port<br>
<br>
in my extension.con<br>
[my-office]<br>
exten=>1401,1,Dial(SIP/1401,60)<br>
exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}@1401)<br>
<br>
but the problem is when I dial the number for the PSTN it's run/dial<br>
on internal extension, from the asterisk guru website it's wrote to<br>
separate the incoming and out going<br>
in sip.conf<br>
[1401]<br>
type=friend<br>
username=1401<br>
secret=1401<br>
host=dynamic<br>
context=my-office-in<br>
insecure=port<br>
<br>
[1401]<br>
type=friend<br>
username=1401<br>
secret=1401<br>
host=dynamic<br>
context=my-office-out<br>
insecure=port<br>
<br>
in extension.conf<br>
[my-office-in]<br>
exten=>1401,1,Dial(SIP/1001,60)<br>
[my-office-out]<br>
exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}@1401)<br>
<br>
but still with this won't work too<br>
My question it's<br>
Is it my configuration true/correct or if there any other way for my problem<br>
I'm using 1 Stage Dialing and the asterisk server and Grandstream<br>
using different IP Address 192.168.101.xxx (for asterisk server) and<br>
192.168.14.xxx (for grandstream gateway)<br>
thank you for helping<br>
--<br>
Regards<br>
Samuel Sappa,<br>
<br>
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</blockquote></div><br></div></div>