[asterisk-users] outbound calls not ringing still

[Olle E. Johansson]

3 sep 2009 kl. 00.27 skrev 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

It's very hard to say much without your configurations at hand.

/O