[Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal

[John Todd]

Try this:

exten => 4001,1,Dial(SIP/gadams,10,r)

I don't know how the syntax you've specified will behave; maybe it 
will work, but it's not any format I've used.  Try the syntax above 
for your Dial line and see if it results in different actions.

Plus, I'm slightly confused as to your dialplan.  When a call comes 
in from your 7960, it is passed to context "ATL".  Now, when you call 
your 7960, where is that call going?  Are you passing your inbound 
and outbound calls both to context "ATL"?   If not, then you're 
missing an important part of the debug (and perhaps having a 
conceptual problem with how call flow works.)

JT


>-----Original Message-----
>From: Adams, Gavin
>Sent: Monday, August 04, 2003 6:10 PM
>To: 'asterisk-users at lists.digium.com'
>Subject: Cisco 7960, SIP, NAT, Voicemal
>
>Hey all,
>
>I've got a couple 79xx phones working peer-to-peer and am now trying to
>work on the voice mail.
>
>In extensions.conf:
>
>[ATL]
>exten => 4001,1,Dial(SIP/gadams)|10
>exten => 4001,2,Voicemail,u4001
>exten => 4001,102,Voicemail,b4001
>
>and the corresponding sip.conf:
>
>[gadams]
>type=friend
>username=gadams
>secret=******
>context=ATL
>host=dynamic
>canreinvite=no
>nat=yes
>mailbox=4001
>
>When this phone is dialed, it doesn't roll over to VM after 10 seconds
>but continues to ring. If the calling party hangs up, the phone
>continues to ring.
>
>However, as a test I changed the |10 to a |10t. At that point dialing
>4001 did indeed roll over to voicemail, but it happened immediately.
>Also, I'm getting the following message during the dial:
>
>WARNING[1133735216]: File chan_sip.c, Line 417 (retrans_pkt): Maximum
>retries exceeded on call 71817d20169869e82ee3d897699393e3 at 63.111.7.161
>for seqno 102 (Request)
>
>Which is tied to the call in question. Any clues?
>
>--- Gavin
>
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