[asterisk-users] Delay before audio starts

[Eric Wieling]

When Answer fixes the issue, the root cause is often NAT (could be firewall) since Answering the call prevents any reinvites.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gerard
Sent: Friday, March 01, 2013 9:33 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Delay before audio starts

I've found a workaround of sorts, If I change my below code to :
 1AAAAAAAAAA => {
         NoOp(${CALLERID(num)});
	 Answer();  // <--------------- add this
         Ringing;
         Set(CHANNEL(musicclass)=none);
         Dial(${OUTBOUND-TRUNKR}/1XXXXXXXXXX,30);
         Voicemail(198,u);
  };

That fixes the issue. It doesn't fix the call forward issue on the phone though. I've made a few extra extensions, one each corresponding to a number he wants to call forward to, if I have him forward to the extensions who then forward to the real number, it works, thanks to adding "Answer()" to the dialplan.

-Gerard


On 02/26/13 13:19, Gerard wrote:
> 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,
> 


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