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?<div>
<br></div><div>Leandro<br><br><div class="gmail_quote">2013/3/1 Gerard <span dir="ltr"><<a href="mailto:gsaraber@rarcoa.com" target="_blank">gsaraber@rarcoa.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I thought it was the re-invites too, but I have it turned off everywhere.<br>
<br>
On 03/01/13 08:36, Eric Wieling wrote:<br>
> When Answer fixes the issue, the root cause is often NAT (could be firewall) since Answering the call prevents any reinvites.<br>
><br>
> -----Original Message-----<br>
> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Gerard<br>
> Sent: Friday, March 01, 2013 9:33 AM<br>
> To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
> Subject: Re: [asterisk-users] Delay before audio starts<br>
><br>
> I've found a workaround of sorts, If I change my below code to :<br>
> 1AAAAAAAAAA => {<br>
> NoOp(${CALLERID(num)});<br>
> Answer(); // <--------------- add this<br>
> Ringing;<br>
> Set(CHANNEL(musicclass)=none);<br>
> Dial(${OUTBOUND-TRUNKR}/1XXXXXXXXXX,30);<br>
> Voicemail(198,u);<br>
> };<br>
><br>
> 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.<br>
><br>
> -Gerard<br>
><br>
><br>
> On 02/26/13 13:19, Gerard wrote:<br>
>> Hi everyone,<br>
>><br>
>> I'm having a hard time figuring this issue out, we just switched from<br>
>> a<br>
>> T1 PRI to a SIP trunk provider and that's when the issue started.<br>
>> Now when someone forwards all calls on their phone to a cellphone,<br>
>> when a customer calls in, Asterisk correctly calls the cellphone and<br>
>> connects the call, but there is a long delay before the audio starts,<br>
>> basically for the first 6-10 seconds of the call there is dead<br>
>> silence, eventually the audio will start and everything works correctly.<br>
>> We never had this problem with the PRI. So I suspect it has something<br>
>> to do with a call coming in as SIP and going out as SIP.<br>
>><br>
>> At first I thought it was a call forwarding issue because I got this<br>
>> message in the console:<br>
>> [Feb 26 12:35:19] NOTICE[1143][C-0000025d]: app_dial.c:958 do_forward:<br>
>> Not accepting call completion offers from call-forward recipient<br>
>> Local/1XXXXXXXXXX@default-00000013;1<br>
>><br>
>> So I put this in my dial plan:<br>
>><br>
>> 1AAAAAAAAAA => {<br>
>> NoOp(${CALLERID(num)});<br>
>> Ringing;<br>
>> Set(CHANNEL(musicclass)=none);<br>
>> Dial(${OUTBOUND-TRUNKR}/1XXXXXXXXXX,30);<br>
>> Voicemail(198,u);<br>
>> };<br>
>><br>
>> So basically as soon as someone calls incoming number AAAAAAAAAA,<br>
>> Asterisk dials phone number XXXXXXXXXX. it's a quick and dirty way to<br>
>> call forward.. and this does the same thing, there's a good 8 second<br>
>> delay before the audio kicks in.<br>
>><br>
>><br>
>> There is a Linux firewall with NAT in the path, but I have no other<br>
>> audio issues, so don't *think* it's a factor.<br>
>> I just upgraded to asterisk 11.2.1.<br>
>><br>
>><br>
>> Asterisk 11.2.1 built by root @ phonesys2 on a i686 running Linux on<br>
>> 2013-02-23 01:40:02 UTC<br>
>><br>
>><br>
>> Any help would be appreciated,<br>
>> Thanks,<br>
>><br>
><br>
><br>
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<br>
--<br>
Gerard Saraber<br>
Network Admin.<br>
Rarcoa, Inc<br>
<a href="tel:%28630%29%20654-2580%20x199" value="+16306542580">(630) 654-2580 x199</a><br>
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</blockquote></div><br></div>