[asterisk-users] Delay before audio starts

[Gerard]

> I think a simple tcpdump of the traffic will show the mystery. It can
> be your provider doing something nasty. Have you tried using some
> other cheap SIP termination? or arrange a fake termination yourself
> on another server?
> 
> Leandro

I thought so too, but it doesn't appear to .

I just bought a door intercom device, set up the extension for it and
it's doing the same thing, when it connects there is a 10 second delay
before the other side can hear my voice.
However watching tcpdump, the audio starts streaming both ways immediately.
Changing the dialplan fixes the issue:
        957 => { // Test door phone
                Answer(); //  <--- this line fixes the problem!
                Dial(SIP/199,20);
                Hangup();
                };

It's an ok workaround for the door intercom, but in the case of the
forwarded calls below, as soon as I Answer() their ringback disappears
and the line goes dead while they wait for our guy to answer the phone.

I may start a separate post about getting ringback to work after Answer();

Thanks for the help by the way.
-Gerard


On 03/01/13 14:34, Leandro Dardini wrote:

> 
> 2013/3/1 Gerard <gsaraber at rarcoa.com>
> 
>> 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,
>>>> 
>>>