[asterisk-users] Delay before audio starts

[Gerard]

On 03/21/13 14:14, Gerard wrote:
>> 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();

As a followup, hold music instead of ringback works fine, so as my
current workaround, I'm using an mp3 of the ringback sound as the hold
music.
Anything is better then a dead line :)


> 
> 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,
>>>>>
>>>>
> 
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-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)