[asterisk-users] asterisk-users Digest, Vol 99, Issue 37

[mitch Johnson]

Chris,

Thanks for answering my message.

I'm currently using version 10.5.1.  I included the error message on the
dial plan to show what errors I was displaying.  The call goes through
after that error message is displayed.  As soon as I hear the phone ring,
it drops my call on the calling phone yet the called phone rings two more
times before being dropped, also.

Are you suggesting I upgrade to a later version of Asterisk?

Thanks

Mitch

Message: 14
Date: Wed, 24 Oct 2012 10:20:37 -0500
From: Christopher Harrington <chris at acsdi.com>
Subject: Re: [asterisk-users] as soon as Phone rings I'm disconnected
        yet phone rings two more times?
To: Asterisk Users Mailing List - Non-Commercial Discussion
        <asterisk-users at lists.digium.com>
Message-ID:
        <CAJLBXEnd9nyyLCpAvwcD+s5jsCL8qS=cYgnUqAuS=Qay8K26CQ at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

On Tue, Oct 23, 2012 at 7:54 PM, Mitchell Johnson
<mitch.johnson7 at gmail.com>wrote:

>
> One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal
> adapter onto my asterisk.
>

What version of Asterisk are you using?

[Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite:
> Call from '' (172.16.200.1:65451) to extension '5000' rejected because
> extension not found in context 'default'.
>

Did you mean to include this notice in your email? It indicates a dialplan
problem.


>     -- Executing [5000 at pstn-incoming:1] Dial("SIP/172.16.200.1-00000006",
> "SIP/5000,20|p") in new stack
>

The pipe has been deprecated in more recent versions of Asterisk, make sure
this isn't related to your issue.

--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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End of asterisk-users Digest, Vol 99, Issue 37
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>
> Message: 12
> Date: Tue, 23 Oct 2012 20:54:50 -0400
> From: Mitchell Johnson <mitch.johnson7 at gmail.com>
> Subject: [asterisk-users] as soon as Phone rings I'm disconnected yet
>         phone rings two more times?
> To: asterisk-users at lists.digium.com
> Message-ID: <C9E439FA-355F-4EAF-A42A-652C2C84412C at gmail.com>
> Content-Type: text/plain; charset="us-ascii"
>
>
> One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal
> adapter onto my asterisk.  I have the 8x8 box connected to the Internet,
> and the phone line connected to an fxo port on a Cisco router:
>
> voice-port 0/2/0
>  connection plar opx 5000
>  caller-id enable
>
> dial-peer voice 200 voip
>  destination-pattern 5...
>  session protocol sipv2
>  session target sip-server
>  codec g711ulaw
> !
> sip-ua
>  sip-server ipv4:172.16.200.212     <------ Asterisk server
>
> When I make a call from the PSTN to the 8x8 box, it does send ring back to
> the asterisk server and the Digium phone does ring.  However, as soon as
> the phone rings the call disconnects yet the actual phone, extension 5000,
> rings two times before it hangs up, also.
>
> The following output is what I see on the Asterisk console:
>
> asterisk*CLI>
>   == Using SIP RTP CoS mark 5
> [Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite:
> Call from '' (172.16.200.1:65451) to extension '5000' rejected because
> extension not found in context 'default'.
>   == Using SIP RTP CoS mark 5
>     -- Executing [5000 at pstn-incoming:1] Dial("SIP/172.16.200.1-00000006",
> "SIP/5000,20|p") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/5000
>     -- SIP/5000-00000007 is ringing
>   == Spawn extension (pstn-incoming, 5000, 1) exited non-zero on
> 'SIP/172.16.200.1-00000006'
>
> The 172.16.200.1 is my router.
>
> sip.conf excerpt:
>
> [5000]
> type=friend
> context=phones
> host=dynamic
> disallow=all
> allow=ulaw
> secret=cisco123
> mailbox=5000 at phones
>
> [172.16.200.1]
> context=pstn-incoming
> type=friend
> host=172.16.200.1
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
>
> [phones]
> exten => 5000,1,Dial(SIP/${EXTEN},20|p)
> exten => 5000,n,Hangup
>
> [pstn-incoming]
> include=phones
>
> Any help would be greatly appreciated,
>
> Thanks,
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