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