[Asterisk-Users] Asterisk and Iconnecthere pause

[Justin Sanders]

I tried out your settings which gave me an interesting difference but still
essentially the same problem.  When I use the exten =>
s,4,Dial(SIP/${ARG1}@iconnect,${ARG2},r) in my extensions file it only rings
once (I'm assuming this is by design) and then when it connects I don't get
that first half second of voice followed by the half second of silence and
then voice, I just get a second of silence and then voice.  Kinda
interesting and may shed some light on the problem for me.

I didn't have the registration line in my sip.conf cause I don't receive
incomming calls with iconnecthere.  I put it in though just for the heck of
it but it didn't change anything.

Anyways thanks for your suggestions, I was beginning to think no one else
would reply :).

Justin

----- Original Message ----- 
From: "John Vogel" <johnv at comcast.net>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, April 29, 2004 11:36 PM
Subject: RE: [Asterisk-Users] Asterisk and Iconnecthere pause


>
> Hi, Justin:
>
> I use IConnectHere with * without problems. I presume you have a
> registration line in your sip.conf? Didn't see it in what you have below.
> Something like the following (I've zapped the real numbers.)
>
> register=12225551234:1234 at sipauth.deltathree.com/12225551234
>
> Also, I have the following in my extensions.conf
>
> [macro-dialiconnect]
> exten => s,1,SetCallerID(${ICONNECT1})
> exten => s,2,SetCIDName(${MYNAME})
> exten => s,3,Background(dial-iconnect)
> exten => s,4,Dial(SIP/${ARG1}@iconnect,${ARG2},r)
>
> [iconnect-forced]
> exten => _61XXXXXXXXXX,1,Macro(dialiconnect,${EXTEN:1},20)
> exten => _61XXXXXXXXXX,2,Playback(vm-goodbye)
> exten => _61XXXXXXXXXX,3,Hangup
>
> Good luck!
> John Vogel
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Justin Sanders
> Sent: Wednesday, April 28, 2004 12:14 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Asterisk and Iconnecthere pause
>
> 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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