[asterisk-users] outbound calls not ringing still

[John A. Sullivan III]

On Wed, 2009-09-02 at 21:31 +0000, Ott Rose wrote:
> i have posted this before but was unable to resolve it. i have some
> new info so i figured i would try again. the trace from bandwidth.com
> are below. they are telling me that the ip that is bold should be our
> ip not bandwidth.com. i have changed every setting that i can see and
> nothing fixes this. Where would i change this at? they cannot tell me.
> 
> INVITE sip:+185993133333 at 216.82.224.202 SIP/2.0
> Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport
> From:"8592192438"<sip:8592192438 at 64.191.130.78>;tag=as0707d433
> To:<sip:+185993133333 at 216.82.224.202>
> Contact:<sip:8592192438 at 216.82.224.202>
> Call-ID: 0f3bdcc9171ef53148e7bab413aea08e at 64.191.130.78
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 02 Sep 2009 21:10:39 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 412
> 
> v=0
> o=root 3831 3831 IN IP4 216.82.224.202
> s=session
> c=IN IP4 216.82.224.202
> t=0 0
> m=audio 17050 RTP/AVP 0 8 3 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 12426 RTP/AVP 31 34 103
> a=rtpmap:31 H261/90000
> a=rtpmap:34 H263/90000
> a=rtpmap:103 h263-1998/90000
> a=sendrecv
> 
<snip>
I know very little about how ringing works but are they providing any
kind of status information to you? Do you need to furnish the ring if
they are not? It seems to me I saw quite a few articles about providing
ring tone, what causes it to fail, and how to work around it.  I assume
you've searched for those already. Just a few thoughts - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsullivan at opensourcedevel.com

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