[Asterisk-Users] Asterisk and Iconnecthere pause

[Justin Sanders]

Yes I am behind a nat, but my * server is set as the dmz.  Do you think something like that would cause that short muted period at the beginning of a call, I thought it just wouldn't communicate if it couldn't talk through the nat.  Also though there was no pause when using the SPA-2000 ata directly to iconnecthere and that was setup behind the nat without port forwarding or anything so I wouldn't think that would be the problem. 

Justin

----- Original Message ----- 
  From: Zac Amsler 
  To: asterisk-users at lists.digium.com 
  Sent: Wednesday, April 28, 2004 5:18 PM
  Subject: Re: [Asterisk-Users] Asterisk and Iconnecthere pause


  Are you actually behind a nat?

  I have * connecting to iconnecthere just fine. 
  I have an external IP for my * server.

  Zac

  On Wed, 2004-04-28 at 14:13, Justin Sanders wrote: 
Hi, I just got a SPA-2000 in and was finally able to complete my asterisk
setup.  I'm making my outgoing calls through iconnecthere from the
asterisk server however I'm running into a problem when placing calls.  I
can connect fine but when the person (or answering machine) picks up I
hear them talk for a about half a second then there is a half a second
pause or muted period and then the audio comes back fine for the rest of
the call.  This is frustrating though cause in that time period is usually
when people say hello when they pickup.  It happens on every call and
there isn't any error messages in the sip debug console.  I think I've
narrowed down the problem to some kind of issue with my setup between the
asterisk server and iconnecthere.  The reason I say that is this doesn't
occur when I call fwd numbers or 1800 numbers through iaxtel but will
occur on those same 1800 numbers through iconnecthere.  Also the problem
isn't my phone or SPA-2000 cause I get the same issue if I redirect a
incoming fwd call (through ipkall) out through iconnecthere.  Another
interesting test is if I connect my SPA-2000 directly to iconnecthere
without going through asterisk I don't have any problems and there is no
pause.  So the issue seems to be something with the way my asterisk setup
is talking with iconnecthere.  My sip.conf is setup to disallow=all,
allow=gsm, allow=ulaw, allow=alaw, however it doesn't make a difference if
I connect with either codec.  Here is my iconnecthere section with
user/pass removed:

[iconnect]
type=friend
secret=
username=
host=natrelay.deltathree.com
dtmfmode=inband
canreinvite=no

All this seems pretty standard.  What settings or debug information should
I look at to fix this issue.  Has anyone else encountered this problem
before.  I'd be happy to supply more information about my setup but since
I'm not sure where the problem would be from I don't know what to post. 
Any help would be appreciated.  Thanks.

--
Justin <justin at sandershosting.com>


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