[Asterisk-Users] Voicemail/Record sending no RTP packets (CNG) back to caller when recording messages
In August and September of last year, there was some discussion of changing the Voicemail and Record applications to send back CNG RTP packets during recording to prevent inbound calls from dropping when they assumed a disconnect after 30 seconds of no RTP frames. Was there any resolution on this issue? I'm bringing up a new SIP provider, and am seeing the same behavior that was reported a year ago with BroadVoice, although on my system the disconnect happens after 10 seconds. -- Ted Cabeen Sr. Systems/Network Administrator Impulse Internet Services ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two Zap Problems with 1.0.2 that appeared at the same time: choppyness and squealing
I've got an * system that is having some real problems with 1.0.2. The biggest problem is that calls going through my T100P get choppy for about 10 seconds every 1 or 2 minutes. Asterisk is running on a debian stable system with current packages. The T100P is plugged into a Adit Channelbank with 8 POTS lines hooked up to the Channelbank. I've watched the vritual memory and CPU status on the * box during the call and the system is totally idle. Looking at the verbose logging in *, there isn't any obvious activity on the console that corresponds with the choppyness. It sounds like this is the echo canceller having problems, but regular phones plugged into the lines sound fine. In early November, this system was running fine on 1.0, but even downgrading back to 1.0 now doesn't fix the problem. Any thoughts? The other problem is related to our legacy PBX system, which we route through * when the * is being tested. To do so, we run the 7 PBX lines into a FXS card on the Channelbank and configure * to bridge the PBX lines to the POTS lines as necessary. The problem is that when the employee talking through * on the PBX hangs up the line, the caller coming in through the POTS line hears a very loud squeal for a second until * hangs up the line. I've run the channelbank with a loopbacked T1 replicating the same problem, and it doesn't happen, so * is definitely involved somehow. Is there a setting to determine how asterisk detects hangups on FXS Zap channels? -- Ted Cabeen http://www.pobox.com/~secabeen[EMAIL PROTECTED] Check Website or Keyserver for PGP/GPG Key BA0349D2 [EMAIL PROTECTED] I have taken all knowledge to be my province. -F. Bacon [EMAIL PROTECTED] Human kind cannot bear very much reality.-T.S.Eliot [EMAIL PROTECTED] ___ 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] ADSI phone vs. IP phone
Andrew Kohlsmith [EMAIL PROTECTED] writes: Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. To quote myself: True, but I don't have to retool my office and install POE switches to use ADSI phones, either. No, I will not put a hub/switch at every desk and then use wall-warts for every phone to get around retooling the office. :-) I'm not going to bastardize my network by placing the equivalent of a 3-port switch or hub at every desk to have the phone system compete with our heavy network users (CAD mostly), and I will fight tooth and nail against having to put a goddamned wall-wart at every station just to power the damned IP phones. :-) Do ADSI phones need wall-warts, or can they drive themselves from the line power? -- Ted Cabeen http://www.pobox.com/~secabeen[EMAIL PROTECTED] Check Website or Keyserver for PGP/GPG Key BA0349D2 [EMAIL PROTECTED] I have taken all knowledge to be my province. -F. Bacon [EMAIL PROTECTED] Human kind cannot bear very much reality.-T.S.Eliot[EMAIL PROTECTED] ___ 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] max queue time; newbie question (fwd)
Martin Pycko [EMAIL PROTECTED] writes: sure, use the 'n' option of the queue and put voicemail app as the next priority Will that work? From my read of the code, the timeout parameter is only checked while the call is being sent to an agent's phone (inside the try_calling function). The timeout doesn't seem to be checked while the user is waiting to get to the head of the queue (inside the wait_our_turn function). Unless the ast_waitfordigit function checks the timeout and I missed it, this solution won't work. Am I reading the code right? On Fri, 9 Jan 2004, Ken Alker wrote: I am just studying Asterisk now and have a question. Is it possible to force anyone who enters a queue into voice mail after they have been in the queue for 30 seconds? -- Ted Cabeen Sr. Systems/Network Administrator Impulse Internet Services ___ 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
