abdoul
2005-Jan-09 15:57 UTC
[Asterisk-Users] call from PSTN, not hearing SIP: 180/RINGING( was call from DID,not hearing RINGTONEs )
Hi, They are using equipments from Cirpack. I don"t understand : Solved it by adding progress_ind setup enable 3 on the voip peer. Where should i add this parameter ? Thanks for your help, AB>Oswaldo Arratia ><mailto:asterisk-users%40lists.digium.com?Subject=[Asterisk-Users] call from PSTN, not hearing SIP: 180/RINGING( was call from DID, not hearing RINGTONEs )&In-Reply-To=Pine.LNX.4.44.0501041640200.23752-100000@cheetah.psv.nu>oarratia >at workersequity.net >Tue Jan 4 09:59:12 CST 2005 > >---------- > >What type of equipment does your DID provider have? >I had the same problem with Cisco and solved it by adding progress_ind >setup enable 3 on the voip peer. > >-----Original Message----- >From: ><lists.digium.com/mailman/listinfo/asterisk-users>asterisk-users-bounces >at lists.digium.com >[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Peter >Svensson >Sent: Tuesday, January 04, 2005 10:43 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] call from PSTN, not hearing SIP: 180/RINGING( >was call from DID, not hearing RINGTONEs ) > >On Tue, 4 Jan 2005, abdoul wrote: > > > Answer found : > > > > Answer first, Ring, Wait, then ... > > > > exten => s,1,Answer > > exten => s,2,Ringing > > exten => s,3,Wait(x) > > exten => s,4,... > >That will cause all callers to you to have to pay even if the call is not >answered. Not very nice. > >You may need to play with the "progressinband" setting and/or adding "r" >when dialing a local extension. > >Peter________________________ a b d o u l aba at gcomnetworks.com SIP: (131) 229-1002 at sip.freeipcall.com -------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20050109/eee59032/attachment.htm