[asterisk-users] Delay before audio starts

[Gerard]

I thought it was the re-invites too, but I have it turned off everywhere.

On 03/01/13 08:36, Eric Wieling wrote:
> 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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-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)