[Asterisk-Users] Asterisk and Iconnecthere pause

John Vogel
2004-04-29T20:36:41Z

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@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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Justin Sanders
Sent: Wednesday, April 28, 2004 12:14 PM
To: asterisk-users@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
secretusernamehost=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@sandershosting.com>


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