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 <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 > > > > -- > > _____________________________________________________________________ > > -- 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 >
-- _____________________________________________________________________ -- 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