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
