[asterisk-users] Delay before audio starts

[Gerard]

Hi everyone,

I'm having a hard time figuring this issue out, we just switched from a
T1 PRI to a SIP trunk provider and that's when the issue started.
Now when someone forwards all calls on their phone to a cellphone, when
a customer calls in, Asterisk correctly calls the cellphone and connects
the call, but there is a long delay before the audio starts, basically
for the first 6-10 seconds of the call there is dead silence, eventually
the audio will start and everything works correctly.
We never had this problem with the PRI. So I suspect it has something to
do with a call coming in as SIP and going out as SIP.

At first I thought it was a call forwarding issue because I got this
message in the console:
[Feb 26 12:35:19] NOTICE[1143][C-0000025d]: app_dial.c:958 do_forward:
Not accepting call completion offers from call-forward recipient
Local/1XXXXXXXXXX at default-00000013;1

So I put this in my dial plan:

1AAAAAAAAAA => {
        NoOp(${CALLERID(num)});
        Ringing;
        Set(CHANNEL(musicclass)=none);
        Dial(${OUTBOUND-TRUNKR}/1XXXXXXXXXX,30);
        Voicemail(198,u);
 };

So basically as soon as someone calls incoming number AAAAAAAAAA,
Asterisk dials phone number XXXXXXXXXX. it's a quick and dirty way to
call forward.. and this does the same thing, there's a good 8 second
delay before the audio kicks in.


There is a Linux firewall with NAT in the path, but I have no other
audio issues, so don't *think* it's a factor.
I just upgraded to asterisk 11.2.1.


Asterisk 11.2.1 built by root @ phonesys2 on a i686 running Linux on
2013-02-23 01:40:02 UTC


Any help would be appreciated,
Thanks,
-- 
Gerard Saraber