hi,<div>exten 000,1.Progress() work in some situation. <br><br><div class="gmail_quote">On Thu, Mar 21, 2013 at 9:30 PM, Gerard <span dir="ltr"><<a href="mailto:gsaraber@rarcoa.com" target="_blank">gsaraber@rarcoa.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">On 03/21/13 14:14, Gerard wrote:<br>
>> I think a simple tcpdump of the traffic will show the mystery. It can<br>
>> be your provider doing something nasty. Have you tried using some<br>
>> other cheap SIP termination? or arrange a fake termination yourself<br>
>> on another server?<br>
>><br>
>> Leandro<br>
><br>
> I thought so too, but it doesn't appear to .<br>
><br>
> I just bought a door intercom device, set up the extension for it and<br>
> it's doing the same thing, when it connects there is a 10 second delay<br>
> before the other side can hear my voice.<br>
> However watching tcpdump, the audio starts streaming both ways immediately.<br>
> Changing the dialplan fixes the issue:<br>
> 957 => { // Test door phone<br>
> Answer(); // <--- this line fixes the problem!<br>
> Dial(SIP/199,20);<br>
> Hangup();<br>
> };<br>
><br>
> It's an ok workaround for the door intercom, but in the case of the<br>
> forwarded calls below, as soon as I Answer() their ringback disappears<br>
> and the line goes dead while they wait for our guy to answer the phone.<br>
><br>
> I may start a separate post about getting ringback to work after Answer();<br>
<br>
As a followup, hold music instead of ringback works fine, so as my<br>
current workaround, I'm using an mp3 of the ringback sound as the hold<br>
music.<br>
Anything is better then a dead line :)<br>
<br>
<br>
><br>
> Thanks for the help by the way.<br>
> -Gerard<br>
><br>
><br>
> On 03/01/13 14:34, Leandro Dardini wrote:<br>
><br>
>><br>
>> 2013/3/1 Gerard <<a href="mailto:gsaraber@rarcoa.com">gsaraber@rarcoa.com</a>><br>
>><br>
>>> I thought it was the re-invites too, but I have it turned off<br>
>>> 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<br>
>>>> be<br>
>>> firewall) since Answering the call prevents any reinvites.<br>
>>>><br>
>>>> -----Original Message----- From:<br>
>>>> <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<br>
>>> <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 To:<br>
>>>> <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a> Subject: Re: [asterisk-users]<br>
>>>> Delay before audio starts<br>
>>>><br>
>>>> I've found a workaround of sorts, If I change my below code to :<br>
>>>> 1AAAAAAAAAA => { NoOp(${CALLERID(num)}); Answer(); //<br>
>>>> <--------------- add this Ringing;<br>
>>>> Set(CHANNEL(musicclass)=none);<br>
>>>> Dial(${OUTBOUND-TRUNKR}/1XXXXXXXXXX,30); Voicemail(198,u); };<br>
>>>><br>
>>>> That fixes the issue. It doesn't fix the call forward issue on<br>
>>>> the phone<br>
>>> though. I've made a few extra extensions, one each corresponding to<br>
>>> a number he wants to call forward to, if I have him forward to the<br>
>>> extensions who then forward to the real number, it works, thanks to<br>
>>> 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<br>
>>>>> switched from a T1 PRI to a SIP trunk provider and that's when<br>
>>>>> the issue started. Now when someone forwards all calls on their<br>
>>>>> phone to a cellphone, when a customer calls in, Asterisk<br>
>>>>> correctly calls the cellphone and connects the call, but there<br>
>>>>> is a long delay before the audio starts, basically for the<br>
>>>>> first 6-10 seconds of the call there is dead silence,<br>
>>>>> eventually the audio will start and everything works<br>
>>>>> correctly. We never had this problem with the PRI. So I suspect<br>
>>>>> it has something to do with a call coming in as SIP and going<br>
>>>>> out as SIP.<br>
>>>>><br>
>>>>> At first I thought it was a call forwarding issue because I got<br>
>>>>> this message in the console: [Feb 26 12:35:19]<br>
>>>>> NOTICE[1143][C-0000025d]: app_dial.c:958 do_forward: Not<br>
>>>>> 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 => { NoOp(${CALLERID(num)}); Ringing;<br>
>>>>> Set(CHANNEL(musicclass)=none);<br>
>>>>> Dial(${OUTBOUND-TRUNKR}/1XXXXXXXXXX,30); Voicemail(198,u); };<br>
>>>>><br>
>>>>> So basically as soon as someone calls incoming number<br>
>>>>> AAAAAAAAAA, Asterisk dials phone number XXXXXXXXXX. it's a<br>
>>>>> quick and dirty way to call forward.. and this does the same<br>
>>>>> thing, there's a good 8 second delay before the audio kicks<br>
>>>>> in.<br>
>>>>><br>
>>>>><br>
>>>>> There is a Linux firewall with NAT in the path, but I have no<br>
>>>>> other audio issues, so don't *think* it's a factor. I just<br>
>>>>> upgraded to asterisk 11.2.1.<br>
>>>>><br>
>>>>><br>
>>>>> Asterisk 11.2.1 built by root @ phonesys2 on a i686 running<br>
>>>>> Linux on 2013-02-23 01:40:02 UTC<br>
>>>>><br>
>>>>><br>
>>>>> Any help would be appreciated, Thanks,<br>
>>>>><br>
>>>><br>
><br>
> --<br>
> _____________________________________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br>
<br>
<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>
<a href="tel:%28630%29%20654-3556" value="+16306543556">(630) 654-3556</a> (fax)<br>
<a href="tel:%28630%29%20915-4122" value="+16309154122">(630) 915-4122</a> (cell)<br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br></div>