[asterisk-users] no outbound calls

[das sandesh]

You have to check and verify the SIP trunk details, as ext to ext works once
the pbx is up, but to call out, it should go through your provider.....so
just recheck your provider's details.
Regards
Sandesh

On Wed, Oct 14, 2009 at 8:24 AM, Ott Rose <sixfourimpala at hotmail.com> wrote:

>  here is the debug from the CLI. I think I know where the problem is I just
> can figure out how to fix it. The IP in the From and To i think is where the
> problem is. When I make an outbound call. i get the message "the call cannot
> be completed as dialed". if i call another ext it works. I posted the debug
> for both calls.
>
>
>
>
>
>
> ==============outbound call===========================
>
> <--- Transmitting (NAT) to 10.0.0.46:5060 --->
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP 10.0.0.46:5060
> ;branch=z9hG4bKfd2143ede5319cf9b.273e4b904b0cc3101;received=10.0.0.46
> From: "ext" <sip:117 at 10.0.0.8 <sip%3A117 at 10.0.0.8>>;tag=9d9e3944ba
> To: "93214545" <sip:93214545 at 10.0.0.8 <sip%3A93214545 at 10.0.0.8>
> >;tag=as290bd498
> Call-ID: 401d30b0a1893e80
> CSeq: 13401 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:99676446 at 10.0.0.8 <sip%3A99676446 at 10.0.0.8>>
> Content-Type: application/sdp
> Content-Length: 254
>
> v=0
> o=root 3609 3609 IN IP4 10.0.0.8
> s=session
> c=IN IP4 10.0.0.8
> t=0 0
> m=audio 14398 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> =====================================================
>
> ================ext to ext===============================
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.46:5060
> ;branch=z9hG4bK280378de3608b5bf1.2576ced30c6198b74;received=10.0.0.46
> From: "ext" <sip:117 at 10.0.0.8 <sip%3A117 at 10.0.0.8>>;tag=d729237fcc
> To: "111" <sip:111 at 10.0.0.8 <sip%3A111 at 10.0.0.8>>;tag=as553ab5e9
> Call-ID: c7cc32657c620790
> CSeq: 8007 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:111 at 10.0.0.8 <sip%3A111 at 10.0.0.8>>
> Content-Type: application/sdp
> Content-Length: 254
>
> v=0
> o=root 3609 3609 IN IP4 10.0.0.8
> s=session
> c=IN IP4 10.0.0.8
> t=0 0
> m=audio 10414 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
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