FW: [Asterisk-Users] RTP timing issues
Dear Sirs, The Asterisk bounty has been updated accordingly. Some info about our environment: Our Asterisk server is logically connected to a Veraz NGN platform through SIP and we are facing two major problems for calls from/to Veraz; When calling from Veraz to any SIP extension, no ringback is generated as Veraz does not generate any RTP packets until Answer supervision. Asterisk can not deliver ringback. Calling to Veraz is problematic as all our interfaces are using Silence compression. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, October 07, 2004 11:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED]; Bart Coppens Subject: Re: [Asterisk-Users] RTP timing issues On Thu, 7 Oct 2004, Bart Coppens wrote: > Some time ago, I announced a bounty to solve the issues with regards to > silence compression (chopped voice) and one way voice. To get this solved, > Asterisk should get the clocking from an internal source in a way that an > ouput stream can be generated without getting any RTP input. > > Now my company is exposing a payment of 1000USD for this bounty. This > payment have to justified through an official invoice. > > Can someone give me an indication if this can be achieved? It can be achieved. Steve _ All about Paris Motor Show 2004 http://motorshow.auto.msn.be ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ringing() doesn't play sound while phone is ringing
try exten => 101,2,Dial(Zap/1,10,r) in stead of exten => 101,2,Dial(Zap/1,10) BC From: "Warren Burstein" <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] To: <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Ringing() doesn't play sound while phone is ringing Date: Wed, 11 Aug 2004 15:22:45 +0400 I have: RedHat 9.0 TDM40B asterisk-0.9.0 compiled from sources zaptel-0.9.1 likewise /etc/zaptel.conf contains fxoks=1-4 loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel => 1-4 /etc/askterisk/extensions.conf contains [general] static=yes writeprotect=yes [phones] exten => 101,1,Ringing() exten => 101,2,Dial(Zap/1,10) exten => 101,3,Congestion I also uncommented the "noload => chan_oss.so" in /etc/asterisk/modules.conf because I don't have a sound card. Other than that, all conf files are the originals from "make samples". But when I dial 101 from Zap/2, Zap/1 rings (and if I pick it up, I can have a conversation with myself), but I don't hear a ringing tone out of Zap/2. I commented out the Dial and Congestion, and then I heard a two ringing tones, a click, and a congestion tone, while the console said: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in context 'phones' I'm guessing that Dial stops Ringing. How do I tell Ringing to continue while Dial is working, and if it isn't stopped by Dial, not to time out after two rings? "show application ringing" doesn't describe any parameters to Ringing() . Thanks. _ Make your own website http://webbuilder.msn.be ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New feature request
Dear all, This feature request is derived from Bug ID 2206 Currently, Asterisk is using the timing of the input stream to reproduce the output stream. This means that when no RTP streams are being sent from the peer Endpoint/GW, Asterisk is unable not generate audio. This approach/limitation can lead to "one way speech" conditions: Some devices don't generate audio until the answer supervision is received from the called. For all these scenarios, no ringback can be presented to the calling party. In cases where the endpoints are using silence compression, the audio from asterisk is chopped. It would be much better to generate audio, even if no RTP is received at all. The clocking should than be taken from an internal timing mechanism that keeps track of the synchronization. A configuration option should exist to choose on the method. Is anybody else interested in such feature request? Cheers, Bart _ Try DVDPost for only 1,00! http://dvd.fr.msn.be/10days1euro.php ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
