[asterisk-users] Delay before audio starts

[Leandro Dardini]

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

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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>    http://lists.digium.com/mailman/listinfo/asterisk-users
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