[asterisk-users] Delay before audio starts

[Asghar Mohammad]

hi,
exten 000,1.Progress() work in some situation.

On Thu, Mar 21, 2013 at 9:30 PM, Gerard <gsaraber at rarcoa.com> wrote:

> 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)
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
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