Hi, Sorry I forgot to mention versions and post files. Asterisk version: pbx:/etc/asterisk# asterisk -rx "core show version" Asterisk 1.4.22 built by root @ coope-pbx on a i686 running Linux on 2008-10-22 09:36:35 UTC
I'm running zaptel 1.4.12.1 and wanpipe 3.3.14. Also tried zaptel 1.4.11 and 1.4.12, and wanpipe 3.2.7.1, and the problem happens on all versions. extensions.conf: http://www.pastebin.ca/1235312 sip.conf: http://www.pastebin.ca/1235317 zapata.conf: http://www.pastebin.ca/1235318 zaptel.conf: http://www.pastebin.ca/1235322 Let me know if I should be posting any other conf files. On Fri, Oct 24, 2008 at 1:27 AM, Lucas Alvarez <[EMAIL PROTECTED]> wrote: > Hi, which version of asterisk are you running? Perhaps if you post your > extensions.conf and others related files you could get more accurate help. > If you answer a ringing phone and you can't answer the call, there you > could have a network or sip config problem, that means that the SIP packet > is not returning to the pbx. > Regards. > > Lucas > > > On Thu, 23 Oct 2008 10:30:29 -0200, Fernando Serto <[EMAIL PROTECTED]> > wrote: > >> Hi, >> >> I've been very puzzled lately. I installed a phone system for a friend >> a few weeks ago, and they're having a problem that I can't get rid of, >> actually 2 problems. Before I go into the problems, let me tell you >> about the setup. It's a pretty small setup with only 4 handsets, all >> Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual >> core, 2GHz) and 512MB Ram. Internet Connection is an ADSL2, with a not >> so reliable ISP in australia. For incoming calls, I had a Digium >> TDM410P with 4xFXO modules and HWEC. Because of these problems, i >> replaced the Digium card with a Sangoma A200D, but it didn't make any >> difference to the problems. All phones are hooked up to a Netgear PoE >> switch. >> >> Almost forgot to mention that this is not my first Asterisk setup, and >> in fact it is my 4th, and I used various SIP handsets before, and also >> different cards (Analog and Digital), so I'm not a total noob. >> >> Let's get to the problems... >> >> 1) Some incoming calls cannot be picked up >> Sometimes, incoming calls, coming through the analog card, cannot be >> picked up. All handsets are set to ring at the same time on incoming >> calls. and most of the time, calls can be answered on any of the >> handsets, but maybe 3 or 4 times a day, all handsets will be ringing, >> and you go to one handset to answer the call, you pick the handset, >> and it doesn't answer the call, it keeps ringing, then you go to >> another handset, and still can't pick up, sometimes, you can even try >> all 4 handsets, and no luck. but, at other times, you can't answer on >> the first handset, but you can on another, and it is totally random. >> but people are pretty pissed off for running around to answer a call. >> and what puzzles me is that you can sit around watching logs for >> hours, and it won't happen, other times, it happens 3 times in a row. >> any ideas? >> >> 2) Delay on outgoing calls via SIP >> People have been saying that when they call people, there's a delay >> for the call to be answered. For example, caller dials a number, >> callee answers the ringing phone, but caller is still listening to a >> ringing tone, and after a few seconds (up to 15 seconds) it sounds >> like the callee has just answered the call, when in fact, he had >> already answered a few seconds before. Problem with this is that some >> callees will hangup before the caller starts talking. These calls are >> going via pennytel, in australia, which seems to be a pretty good VOIP >> provider around here, and I've been using it on other setups and never >> had these issues. >> >> Well, sorry for the long first post, but I would really appreciate any >> suggestions you have. >> >> Cheers, >> Fernando >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
