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 <gsara...@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-boun...@lists.digium.com [mailto: >>> asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard >>>> Sent: Friday, March 01, 2013 9:33 AM To: >>>> asterisk-users@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@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, >>>>> >>>> > > -- > _____________________________________________________________________ > -- 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users